Compare commits

..

87 Commits

Author SHA1 Message Date
Michael Niedermayer
bc839fb39d Changelog: Update for the last 3 commits
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-10 21:20:05 +01:00
Michael Niedermayer
1fab842fbb avcodec/vp9_superframe_split_bsf: Fix integer overflow in frame_size/total_size checks
Fixes: signed integer overflow: -1698586465 + -551542752 cannot be represented in type 'int'
Fixes: 4490/clusterfuzz-testcase-minimized-5210014592532480

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit eaff5fcb7c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-10 20:25:14 +01:00
Michael Niedermayer
60d250386b avcodec/amrwbdec: Fix division by 0 in voice_factor()
The added value matches "Digital cellular telecommunications system (Phase 2+) (GSM); Universal Mobile Telecommunications System (UMTS); LTE; Extended Adaptive Multi-Rate - Wideband (AMR-WB+) codec; Floating-point ANSI-C code (3GPP TS 26.304 version 14.0.0 Release 14)
Extended Adaptive Multi-Rate - Wideband (AMR-WB+) codec; Floating-point ANSI-C code"

Fixes: runtime error: division by zero
Fixes: 4415/clusterfuzz-testcase-minimized-4677752314658816

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 1d0817d56b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-10 20:25:14 +01:00
Michael Niedermayer
c5fd23879a avformat/utils: Fix warning: ISO C90 forbids mixed declarations and code
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-10 20:25:14 +01:00
James Cowgill
d8d1689f92 avcodec/decode: reset codec on receiving packet after EOF in compat_decode
In commit 061a0c14bb ("decode: restructure the core decoding code"), the
deprecated avcodec_decode_* APIs were reworked so that they called into the
new avcodec_send_packet / avcodec_receive_frame API. This had the side effect
of prohibiting sending new packets containing data after a drain
packet, but in previous versions of FFmpeg this "worked" and some
applications relied on it.

To restore some compatibility, reset the codec if we receive a new non-drain
packet using the old API after draining has completed. While this does
not give the same behaviour as the old API did, in the majority of cases
it works and it does not require changes to any other part of the decoding
code.

Fixes ticket #6775
Signed-off-by: James Cowgill <jcowgill@debian.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 02ba4b91b5)
2017-12-09 21:40:47 +01:00
Michael Niedermayer
c741095eec Update for 3.4.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-08 10:22:30 +01:00
Michael Niedermayer
b2169c8bcc avcodec/diracdsp: Fix integer overflow in PUT_SIGNED_RECT_CLAMPED()
Fixes: runtime error: signed integer overflow: 2147483646 + 2048 cannot be represented in type 'int'
Fixes: 4479/clusterfuzz-testcase-minimized-6529894147162112

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 610dd74502)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
3a6140e4cf avcodec/dirac_dwt: Fix integer overflows in COMPOSE_DAUB97*
Fixes: 4478/clusterfuzz-testcase-minimized-4752113767809024
Fixes: runtime error: signed integer overflow: -2147483626 + -319489 cannot be represented in type 'int'

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5e9a13a5a3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Nikolas Bowe
a749f4864e avcodec/extract_extradata_bsf: Fix leak discovered via fuzzing
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5a412a5c3c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
c147aefc3e avcodec/vorbis: Fix another 1 << 31 > int32_t::max() with 1u.
Didn't notice this one when 9648cc6d was landed.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 95bacb521a)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
23319f7764 avcodec/vorbis: 1 << 31 > int32_t::max(), so use 1u << 31 instead.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9648cc6d7f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
36db62ca98 avformat/utils: Prevent undefined shift with wrap_bits > 64.
2LL << (wrap_bits=64 - 1) does not fit in int64_t; change the
code to use a uint64_t (2ULL) and add an av_assert2() to
ensure wrap_bits <= 64.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 03fbc0daa7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
17f05ff656 avcodec/j2kenc: Fix out of array access in encode_cblk()
Fixes: 4427/clusterfuzz-testcase-minimized-5106919271301120

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0674087004)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
0ccbbf034d avcodec/hevcdsp_template: Fix undefined shift in put_hevc_epel_bi_w_h()
Fixes: runtime error: left shift of negative value -127
Fixes: 4397/clusterfuzz-testcase-minimized-4779061080489984

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0409d33311)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
John Stebbins
f7357facd8 lavf/mov: fix huge alloc in mov_read_ctts
An invalid file may cause huge alloc.  Delay expansion of ctts entries
until the number of samples is known in mov_build_index.

Fixes: 23

Found-by: zhao dongzhuo, AD-lab of Venustech
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2d015d3bf9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
ed87667bd3 avcodec/mlpdsp: Fix signed integer overflow, 2nd try
The outputted bits should match what is used in the lossless check

Fixes: runtime error: signed integer overflow: -538697856 * 256 cannot be represented in type 'int'
Fixes: 4326/clusterfuzz-testcase-minimized-5689449645080576

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 97c00edaa0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
983d119c9b avcodec/h264idct_template: Fix integer overflow in ff_h264_idct8_add
Fixes: signed integer overflow: 452986184 - -2113885312 cannot be represented in type 'int'
Fixes: 4196/clusterfuzz-testcase-minimized-5580648594014208

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9cc926da7d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
e56f691283 avcodec/kgv1dec: Check that there is enough input for maximum RLE compression
Fixes: Timeout
Fixes: 4271/clusterfuzz-testcase-4676667768307712

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 3aad94bf2b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
3ae71b648a avformat/aacdec: Fix leak in adts_aac_read_packet()
Fixes: chromium-773637/clusterfuzz-testcase-minimized-6418078673141760

Found-by: ossfuzz/chromium
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2779d33ed9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
f2f0273588 avcodec/dirac_dwt: Fix integer overflow in COMPOSE_FIDELITYi*
Fixes: runtime error: signed integer overflow: -2143827186 - 7404944 cannot be represented in type 'int'
Fixes: 4354/clusterfuzz-testcase-minimized-4671122764201984

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2b6964f764)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
de20dad15e avcodec/sbrdsp_fixed: Fix integer overflow
Fixes: signed integer overflow: 2147483598 + 64 cannot be represented in type 'int'
Fixes: 4337/clusterfuzz-testcase-minimized-6192658616680448

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 12a511f2c2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
1549890035 avcodec/mpeg4videodec: Check also for negative versions in the validity check
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0e7865ce41)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
35c7a1df8a Close ogg stream upon error when using AV_EF_EXPLODE.
Without this there can be multiple memory leaks for unrecognized
ogg streams.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit bce8fc0754)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
f8fcb6bbf0 Fix undefined shift on assumed 8-bit input.
decode_user_data() attempts to create an integer |build|
value with 8 bits of spacing for 3 components. However
each component is an int32_t, so shifting each component
is undefined for values outside of the 8 bit range.

This patch simply clamps input to 8-bits per component
and prints out a warning that the values were clamped.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7010dd98b5)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
50c93ce5ef Use ff_thread_once for fixed, float table init.
These tables are static so they should only be initialized once
instead of on every call to ff_mpadsp_init().

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 5eaaffaf64)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Dale Curtis
9a00ce0ff8 Fix leak of frame_duration_buffer in mov_fix_index().
Should be unconditionally freed at the end of mov_fix_index() in
case it hasn't been used during the fix up.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Sasi Inguva <isasi-at-google.com@ffmpeg.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit d073be2291)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Jacob Trimble
8aabc4fdb5 avformat/mov: Propagate errors in mov_switch_root.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2d9cf3bf16)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
2e58db3db0 avcodec/hevcdsp_template: Fix invalid shift in put_hevc_epel_bi_w_v()
Fixes: runtime error: left shift of negative value -255
Fixes: 4037/clusterfuzz-testcase-minimized-5290998163832832

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7d88586e47)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
4942de6f93 avcodec/mlpdsp: Fix undefined shift ff_mlp_pack_output()
Fixes: runtime error: left shift of negative value -7862264
Fixes: 4074/clusterfuzz-testcase-minimized-4516104123711488

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4f7f70738e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
cc9d1bb839 avcodec/zmbv: Check that the buffer is large enough for mvec
Fixes: Timeout
Fixes: 4143/clusterfuzz-testcase-4736864637419520

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2ab9568a2c)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
0ba93614cf avcodec/dirac_dwt: Fix integer overflow in COMPOSE_DD137iL0()
Fixes: 4035/clusterfuzz-testcase-minimized-6479308925173760
Fixes: runtime error: signed integer overflow: 9 * 402653183 cannot be represented in type 'int'

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 73964680d7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
ecf2755a41 avcodec/wmv2dec: Check end of bitstream in parse_mb_skip() and ff_wmv2_decode_mb()
Fixes: Timeout
Fixes: 3200/clusterfuzz-testcase-5750022136135680

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 65e0a7c473)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
23d5f15b42 avcodec/snowdec: Check for remaining bitstream in decode_blocks()
Fixes: Timeout
Fixes: 3142/clusterfuzz-testcase-5007853163118592

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4527ec2216)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
77cfc820cf avcodec/snowdec: Check intra block dc differences.
Fixes: Timeout
Fixes: 3142/clusterfuzz-testcase-5007853163118592

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c3b9bbcc6e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Fredrik Hubinette
53715eb13e avformat/mov: Check size of STSC allocation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a6fdd75fe6)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
7b16eacf81 avcodec/vc2enc: Clear coef_buf on allocation
Fixes: Use of uninitialized memory
Fixes: assertion failure

Reviewed-by: <atomnuker>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 6d00905f81)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
d25736dc87 avcodec/h264dec: Fix potential array overread
add padding before scantable arrays

See: 522d850e68

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 380b48fb9f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
6ccf19198b avcodec/x86/mpegvideodsp: Fix signedness bug in need_emu
Fixes: out of array read
Fixes: 3516/attachment-311488.dat

Found-by: Insu Yun, Georgia Tech.
Tested-by: wuninsu@gmail.com
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 58cf31cee7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
44fb120112 avcodec/aacpsdsp_template: Fix integer overflows in ps_decorrelate_c()
Fixes: runtime error: signed integer overflow: 1939661764 - -454942263 cannot be represented in type 'int'
Fixes: 3191/clusterfuzz-testcase-minimized-5688798451073024

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2afe05402f)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
db82e4f1e0 avcodec/aacdec_fixed: Fix undefined shift
Fixes: runtime error: left shift of negative value -801112064
Fixes: 3492/clusterfuzz-testcase-minimized-5784775283441664

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fca198fb5b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
168ee58255 avcodec/mdct_*: Fix integer overflow in addition in RESCALE()
Fixes: runtime error: signed integer overflow: 1219998458 - -1469874012 cannot be represented in type 'int'
Fixes: 3443/clusterfuzz-testcase-minimized-5369987105554432

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 770c934fa1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
3a143bfa19 avcodec/snowdec: Fix integer overflow in header parsing
Fixes: 3984/clusterfuzz-testcase-minimized-5265759929368576
Fixes: runtime error: signed integer overflow: -1085585801 + -1094995529 cannot be represented in type 'int'

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit c897a92858)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
ed87b8b61f avcodec/cngdec: Fix integer clipping
Fixes: runtime error: value -36211.7 is outside the range of representable values of type 'short'
Fixes: 2992/clusterfuzz-testcase-6649611793989632

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 51090133b3)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
87f39642f3 avcodec/sbrdsp_fixed: Fix integer overflow in shift in sbr_hf_g_filt_c()
Fixes: runtime error: shift exponent 66 is too large for 64-bit type 'long long'
Fixes: 3642/clusterfuzz-testcase-minimized-5443853801750528

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 981e99ab99)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
8ec1ff14fe avcodec/aacsbr_fixed: Fix division by zero in sbr_gain_calc()
Fixes: 3642/clusterfuzz-testcase-minimized-5443853801750528

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 7d1dec4668)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
3f2be02b4d avutil/softfloat: Add FLOAT_MIN
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e34fe61bf4)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
56ce961cc3 avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
Fixes: runtime error: signed integer overflow: -503316480 + -2013265038 cannot be represented in type 'int'
Fixes: 3805/clusterfuzz-testcase-minimized-6578427831255040

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e131b8cedb)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
3ca4f1868d avcodec/xan: Check for bitstream end in xan_huffman_decode()
Fixes: Timeout
Fixes: 3707/clusterfuzz-testcase-6465922706440192

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 4b51437dcc)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
0ee2cb866c avcodec/exr: fix undefined shift in pxr24_uncompress()
Fixes: runtime error: left shift of 255 by 24 places cannot be represented in type 'int'
Fixes: 3787/clusterfuzz-testcase-minimized-5728764920070144

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 66f0c958bf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Luca Barbato
78b8aeee58 avformat: Free the internal codec context at the end
Avoid a use after free in avformat_find_stream_info.

(cherry picked from commit 9e4a5eb51b)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
66e65e0a68 avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
Fixes: runtime error: signed integer overflow: 924846844 + 1457520640 cannot be represented in type 'int'
Fixes: 3416/clusterfuzz-testcase-minimized-6125587682820096

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2b739e1cb8)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
6be60aedcb avcodec/xan: Improve overlapping check
Fixes: memcpy-param-overlap
Fixes: 3612/clusterfuzz-testcase-minimized-6393461273001984

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit e8fafef1db)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
dccead84c6 avcodec/aacdec_fixed: Fix integer overflow in apply_dependent_coupling_fixed()
Fixes: runtime error: signed integer overflow: 623487 * 536870912 cannot be represented in type 'int'
Fixes: 3594/clusterfuzz-testcase-minimized-4650622935629824

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 41d96af2a7)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
b3bdb0ddc1 avcodec/aacdec_fixed: Fix integer overflow in predict()
Fixes: runtime error: signed integer overflow: -2110708110 + -82837504 cannot be represented in type 'int'
Fixes: 3547/clusterfuzz-testcase-minimized-6009386439802880

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0976752420)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
7a23220bf9 avcodec/jpeglsdec: Check for end of bitstream in ls_decode_line()
Fixes: 1773/clusterfuzz-testcase-minimized-4832523987189760

Fixes: Timeout

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit f80224ed19)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
1c931d5ab9 avcodec/jpeglsdec: Check ilv for being a supported value
Fixes: 1773/clusterfuzz-testcase-minimized-4832523987189760

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit fe533628b9)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
7ff156b112 tests/ffserver.regression.ref: update checksums to what ffserver currently produces
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 431eccd61e)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
Michael Niedermayer
561e276899 ffserver: Fix off by 1 error in path
Code suggested by ubitux

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 617f0c65e1)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-12-07 23:38:06 +01:00
James Almer
bcfbcbec48 avcodec/proresdec: align dequantization matrix buffers
Should fix ticket #6838

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit f399172d6e)
2017-12-01 01:27:24 -03:00
James Almer
2940b3e17c avformat/matroskaenc: add missing allocation failure checks for stream durations
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 5f67073b4c)
2017-12-01 01:26:15 -03:00
James Almer
8d51090dcb avformat/matroskaenc: actually enforce the stream limit
Prevents out of array accesses. Adressess ticket #6873

Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 9d464dc3fc)
2017-12-01 01:25:45 -03:00
Jacob Trimble
5ab992cd38 configure: Fix dependencies of aac_at decoder.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 83ecdc9a92)
2017-12-01 01:24:25 -03:00
Dale Curtis
ceed79323c Don't manipulate duration when it's AV_NOPTS_VALUE.
This leads to signed integer overflow.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit c5fd57f483)
2017-12-01 01:24:17 -03:00
Michael Roitzsch
752659ff1e lavfi/af_pan: fix sign handling in channel coefficient parser
When a channel formula ends with a subtraction, the next formula will
otherwise have its first coefficient negated.

(cherry picked from commit 4f4e19914d)
2017-11-21 14:04:10 +01:00
Steven Liu
67c0793835 avformat/hlsenc: write fmp4 init header after first AV frame
fix ticket id: 6825

Signed-off-by: Steven Liu <lq@onvideo.cn>
Tested-by: Aman Gupta <aman@tmm1.net>
2017-11-15 17:53:42 -08:00
Timo Rothenpieler
62e99f026a avformat/hlsenc: allocate space for terminating null
Fixes CID #1420394
2017-11-15 17:53:38 -08:00
Steven Liu
e3c09fb986 avformat/hlsenc: reindent hlsenc code
Signed-off-by: Steven Liu <lq@onvideo.cn>
2017-11-15 17:53:35 -08:00
Steven Liu
fac3cfb6c1 avformat/hlsenc: check hls segment mode for ignore the init filename
ignore the fmp4_init_filename when in normal hls segment mode

Signed-off-by: Steven Liu <lq@onvideo.cn>
2017-11-15 17:53:27 -08:00
Steven Liu
9ccb6de56c avformat/hlsenc: reindent hlsenc code
Signed-off-by: Steven Liu <lq@onvideo.cn>
2017-11-15 17:53:21 -08:00
Steven Liu
6ad4d3c92f avformat/hlsenc: fix missing first segment bug in fmp4 mode
fix ticket id: #6776
fix code logic error, need not check first segment.

Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
2017-11-15 17:53:18 -08:00
Steven Liu
d55794fafc avformat/hlsenc: fix base_output_dirname is null when basename_size is 0 bug
fix ticket id: #6777
when use argument hls_segment_filename, the basename_size will be 0

Signed-off-by: Steven Liu <lq@onvideo.cn>
2017-11-15 17:53:14 -08:00
Marton Balint
88a6fca74d ffplay: use SDL2 audio API
It allows us to specify what kind of audio parameter changes are allowed.

Should fix ticket #6721.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit d68a557df4)
2017-11-12 21:00:26 +01:00
Marton Balint
46aa734646 ffplay: only use hardware accelerated SDL texture formats
Typically only a small subset of the SDL texture formats are supported directly
by the SDL renderer drivers, the rest is software emulated. It's better if
libswscale does the format conversion to a hardware-accelerated texture format
instead of SDL.

This should fix video render slowdowns with some texture formats after
3bd2228d05.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 415038f2bd)
2017-11-12 20:59:42 +01:00
Marton Balint
0158fd5276 ffplay: create the window and the renderer before starting playback
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 84d31e2475)
2017-11-12 20:59:27 +01:00
Marton Balint
0ca0ec26a6 ffmpeg: always init output stream before reaping filters
Otherwise the frame size of the codec is not set in the buffersink.

Fixes ticket #6603 and the following simpler case:

ffmpeg -c aac -filter_complex "sine=d=0.1,asetnsamples=1025" out.aac

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit f4090940bd)
2017-11-12 20:58:24 +01:00
Rostislav Pehlivanov
a94cb36ab2 vc2enc_dwt: pad the temporary buffer by the slice size
Since non-Haar wavelets need to look into pixels outside the frame, we
need to pad the buffer. The old factor of two seemed to be a workaround
that fact and only padded to the left and bottom. This correctly pads
by the slice size and as such reduces memory usage and potential
exploits.
Reported by Liu Bingchang.

Ideally, there should be no temporary buffer but the encoder is designed
to deinterleave the coefficients into the classical wavelet structure
with the lower frequency values in the top left corner.

Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
(cherry picked from commit 3228ac730c)
2017-11-09 02:10:56 +00:00
Martin Storsjö
587fadaef1 lavu/arm: Check for have_vfp_vm instead of !have_vfpv3 for float_dsp_vfp
This was missed in e754c8e8 / e2710e790c since those functions
weren't exercised by checkasm.

Fixes ticket #6766.
(cherry picked from commit f1fd12ef85)
2017-10-23 13:31:37 +02:00
Mark Thompson
01e291a592 hwcontext_vaapi: Remove use of vaExportSurfaceHandle()
It is not present in libva 2.0.
2017-10-15 12:45:15 +01:00
Michael Niedermayer
03351cce88 Update versions for 3.4 release
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-15 02:30:15 +02:00
Michael Niedermayer
46abeb1c32 avcodec/snowdec: Check mv_scale
Fixes: runtime error: signed integer overflow: 2 * -1094995530 cannot be represented in type 'int'
Fixes: 3512/clusterfuzz-testcase-minimized-4812747210489856

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 393d6fc739)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-15 01:43:23 +02:00
Michael Niedermayer
35e36046f1 avcodec/pafvideo: Check for bitstream end in decode_0()
Fixes: Timeout
Fixes: 3529/clusterfuzz-testcase-5057068371279872

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9c85329cd0)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-15 01:43:23 +02:00
Sasi Inguva
8500de89ea ffmpeg.c: Fallback to duration_dts, when duration_pts can't be determined.
This is required for FLV files, for which duration_pts comes out to be zero.

Signed-off-by: Sasi Inguva <isasi@google.com>
Reviewed-by: Thomas Mundt <tmundt75@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 2b006ccf83)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-15 01:43:23 +02:00
Ivan Kalvachev
a11a18b284 Fix visual glitch with XvMC, caused by wrong idct permutation.
In the past XvMC forced simple_idct since
it was using FF_IDCT_PERM_NONE.
However now we have SIMD variants of simple_idct that
are using FF_IDCT_PERM_TRANSPOSE and if they are selected
XvMC would get coefficients in the wrong order.

The patch creates new FF_IDCT_NONE that
is used only for this kind of hardware decoding
and that fallbacks to the old C only simple idct.

Signed-off-by: Ivan Kalvachev <ikalvachev@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 9054439bad)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-15 01:43:23 +02:00
James Almer
7deb7e6acd configure: force erroring out in check_disable_warning() if an option doesn't exists
Should prevent some options from being added to cflags when they
don't exist and the compiler only warns about it.

Reviewd-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit ad56e8057d)
2017-10-13 20:57:38 -03:00
Ivan Kalvachev
7fb85ad360 Fix crash if av_vdpau_bind_context() is not used.
The public functions av_alloc_vdpaucontext() and
av_vdpau_alloc_context() are allocating AVVDPAUContext
structure that is supposed to be placed in avctx->hwaccel_context.

However the rest of libavcodec/vdpau.c uses avctx->hwaccel_context
as struct VDPAUHWContext, that is bigger and does contain
AVVDPAUContext as first member.

The usage includes write to the new variables in the bigger stuct,
without checking for block size.

Fix by always allocating the bigger structure.

Signed-off-by: Ivan Kalvachev <ikalvachev@gmail.com>
(cherry picked from commit 3a6ded7cfc)
2017-10-13 00:14:54 +02:00
Marton Balint
c8642473e0 configure: remove libdl dependency from libndi_newtek
We are not using dynamic loading for libndi.

Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 58143b15ad)
2017-10-11 22:50:51 +02:00
Michael Niedermayer
b1ec41a64f add release notes based on release 3.3
Name suggestion was from Helmut K. C. Tessarek

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 07e7ebf52d)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-10-11 03:11:33 +02:00
3954 changed files with 161128 additions and 391468 deletions

2
.gitignore vendored
View File

@@ -29,6 +29,7 @@
/ffmpeg
/ffplay
/ffprobe
/ffserver
/config.asm
/config.h
/coverage.info
@@ -36,4 +37,3 @@
/lcov/
/src
/mapfile
/tools/python/__pycache__/

View File

@@ -1,25 +0,0 @@
<james.darnley@gmail.com> <jdarnley@obe.tv>
<jeebjp@gmail.com> <jan.ekstrom@aminocom.com>
<sw@jkqxz.net> <mrt@jkqxz.net>
<u@pkh.me> <cboesch@gopro.com>
<zhilizhao@tencent.com> <quinkblack@foxmail.com>
<zhilizhao@tencent.com> <wantlamy@gmail.com>
<modmaker@google.com> <modmaker-at-google.com@ffmpeg.org>
<stebbins@jetheaddev.com> <jstebbins@jetheaddev.com>
<barryjzhao@tencent.com> <mypopydev@gmail.com>
<barryjzhao@tencent.com> <jun.zhao@intel.com>
<josh@itanimul.li> <joshdk@obe.tv>
<michael@niedermayer.cc> <michaelni@gmx.at>
<linjie.justin.fu@gmail.com> <linjie.fu@intel.com>
<linjie.justin.fu@gmail.com> <fulinjie@zju.edu.cn>
<ceffmpeg@gmail.com> <cehoyos@ag.or.at>
<ceffmpeg@gmail.com> <cehoyos@rainbow.studorg.tuwien.ac.at>
<ffmpeg@gyani.pro> <gyandoshi@gmail.com>
<atomnuker@gmail.com> <rpehlivanov@obe.tv>
<lizhong1008@gmail.com> <zhong.li@intel.com>
<lizhong1008@gmail.com> <zhongli_dev@126.com>
<andreas.rheinhardt@gmail.com> <andreas.rheinhardt@googlemail.com>
rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
<thilo.borgmann@mail.de> <thilo.borgmann@googlemail.com>
<liuqi05@kuaishou.com> <lq@chinaffmpeg.org>
<ruiling.song83@gmail.com> <ruiling.song@intel.com>

View File

@@ -11,15 +11,11 @@ addons:
compiler:
- clang
- gcc
matrix:
exclude:
- os: osx
compiler: gcc
cache:
directories:
- ffmpeg-samples
before_install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update; fi
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew update --all; fi
install:
- if [ "$TRAVIS_OS_NAME" == "osx" ]; then brew install nasm; fi
script:

373
Changelog
View File

@@ -1,303 +1,82 @@
Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
- AudioToolbox output device
- MacCaption demuxer
- PGX decoder
- chromanr video filter
- VDPAU accelerated HEVC 10/12bit decoding
- ADPCM IMA Ubisoft APM encoder
- Rayman 2 APM muxer
- AV1 encoding support SVT-AV1
- Cineform HD encoder
- ADPCM Argonaut Games encoder
- Argonaut Games ASF muxer
- AV1 Low overhead bitstream format demuxer
- RPZA video encoder
- ADPCM IMA MOFLEX decoder
- MobiClip FastAudio decoder
- MobiClip video decoder
- MOFLEX demuxer
- MODS demuxer
- PhotoCD decoder
- MCA demuxer
- AV1 decoder (Hardware acceleration used only)
- SVS demuxer
- Argonaut Games BRP demuxer
- DAT demuxer
- aax demuxer
- IPU decoder, parser and demuxer
- Intel QSV-accelerated AV1 decoding
- Argonaut Games Video decoder
- libwavpack encoder removed
- ACE demuxer
- AVS3 demuxer
- AVS3 video decoder via libuavs3d
- Cintel RAW decoder
- VDPAU accelerated VP9 10/12bit decoding
- afreqshift and aphaseshift filters
- High Voltage Software ADPCM encoder
- LEGO Racers ALP (.tun & .pcm) muxer
- AV1 VAAPI decoder
- adenorm filter
- ADPCM IMA AMV encoder
- AMV muxer
- NVDEC AV1 hwaccel
- DXVA2/D3D11VA hardware accelerated AV1 decoding
- speechnorm filter
- SpeedHQ encoder
- asupercut filter
- asubcut filter
- Microsoft Paint (MSP) version 2 decoder
- Microsoft Paint (MSP) demuxer
- AV1 monochrome encoding support via libaom >= 2.0.1
- asuperpass and asuperstop filter
- shufflepixels filter
- tmidequalizer filter
- estdif filter
- epx filter
- Dolby E parser
- shear filter
- kirsch filter
- colortemperature filter
- colorcontrast filter
- PFM encoder
- colorcorrect filter
- binka demuxer
- XBM parser
- xbm_pipe demuxer
- colorize filter
- CRI parser
- aexciter audio filter
- exposure video filter
- monochrome video filter
- setts bitstream filter
- vif video filter
- OpenEXR image encoder
- Simbiosis IMX decoder
- Simbiosis IMX demuxer
- Digital Pictures SGA demuxer and decoders
- TTML subtitle encoder and muxer
- identity video filter
- msad video filter
- gophers protocol
- RIST protocol via librist
version 4.3:
- v360 filter
- Intel QSV-accelerated MJPEG decoding
- Intel QSV-accelerated VP9 decoding
- Support for TrueHD in mp4
- Support AMD AMF encoder on Linux (via Vulkan)
- IMM5 video decoder
- ZeroMQ protocol
- support Sipro ACELP.KELVIN decoding
- streamhash muxer
- sierpinski video source
- scroll video filter
- photosensitivity filter
- anlms filter
- arnndn filter
- bilateral filter
- maskedmin and maskedmax filters
- VDPAU VP9 hwaccel
- median filter
- QSV-accelerated VP9 encoding
- AV1 encoding support via librav1e
- AV1 frame merge bitstream filter
- AV1 Annex B demuxer
- axcorrelate filter
- mvdv decoder
- mvha decoder
- MPEG-H 3D Audio support in mp4
- thistogram filter
- freezeframes filter
- Argonaut Games ADPCM decoder
- Argonaut Games ASF demuxer
- xfade video filter
- xfade_opencl filter
- afirsrc audio filter source
- pad_opencl filter
- Simon & Schuster Interactive ADPCM decoder
- Real War KVAG demuxer
- CDToons video decoder
- siren audio decoder
- Rayman 2 ADPCM decoder
- Rayman 2 APM demuxer
- cas video filter
- High Voltage Software ADPCM decoder
- LEGO Racers ALP (.tun & .pcm) demuxer
- AMQP 0-9-1 protocol (RabbitMQ)
- Vulkan support
- avgblur_vulkan, overlay_vulkan, scale_vulkan and chromaber_vulkan filters
- ADPCM IMA MTF decoder
- FWSE demuxer
- DERF DPCM decoder
- DERF demuxer
- CRI HCA decoder
- CRI HCA demuxer
- overlay_cuda filter
- switch from AvxSynth to AviSynth+ on Linux
- mv30 decoder
- Expanded styling support for 3GPP Timed Text Subtitles (movtext)
- WebP parser
- tmedian filter
- maskedthreshold filter
- Support for muxing pcm and pgs in m2ts
- Cunning Developments ADPCM decoder
- asubboost filter
- Pro Pinball Series Soundbank demuxer
- pcm_rechunk bitstream filter
- scdet filter
- NotchLC decoder
- gradients source video filter
- MediaFoundation encoder wrapper
- untile filter
- Simon & Schuster Interactive ADPCM encoder
- PFM decoder
- dblur video filter
- Real War KVAG muxer
version 4.2:
- tpad filter
- AV1 decoding support through libdav1d
- dedot filter
- chromashift and rgbashift filters
- freezedetect filter
- truehd_core bitstream filter
- dhav demuxer
- PCM-DVD encoder
- GIF parser
- vividas demuxer
- hymt decoder
- anlmdn filter
- maskfun filter
- hcom demuxer and decoder
- ARBC decoder
- libaribb24 based ARIB STD-B24 caption support (profiles A and C)
- Support decoding of HEVC 4:4:4 content in nvdec and cuviddec
- removed libndi-newtek
- agm decoder
- KUX demuxer
- AV1 frame split bitstream filter
- lscr decoder
- lagfun filter
- asoftclip filter
- Support decoding of HEVC 4:4:4 content in vdpau
- colorhold filter
- xmedian filter
- asr filter
- showspatial multimedia filter
- VP4 video decoder
- IFV demuxer
- derain filter
- deesser filter
- mov muxer writes tracks with unspecified language instead of English by default
- add support for using clang to compile CUDA kernels
version 4.1:
- deblock filter
- tmix filter
- amplify filter
- fftdnoiz filter
- aderivative and aintegral audio filters
- pal75bars and pal100bars video filter sources
- support mbedTLS based TLS
- adeclick filter
- adeclip filter
- libtensorflow backend for DNN based filters like srcnn
- vc1 decoder is now bit-exact
- ATRAC9 decoder
- lensfun wrapper filter
- colorconstancy filter
- AVS2 video decoder via libdavs2
- IMM4 video decoder
- Brooktree ProSumer video decoder
- MatchWare Screen Capture Codec decoder
- WinCam Motion Video decoder
- 1D LUT filter (lut1d)
- RemotelyAnywhere Screen Capture decoder
- cue and acue filters
- support for AV1 in MP4
- transpose_npp filter
- AVS2 video encoder via libxavs2
- amultiply filter
- Block-Matching 3d (bm3d) denoising filter
- acrossover filter
- ilbc decoder
- audio denoiser as afftdn filter
- AV1 parser
- SER demuxer
- sinc audio filter source
- chromahold filter
- setparams filter
- vibrance filter
- decoding S12M timecode in h264
- xstack filter
- pcm vidc decoder and encoder
- (a)graphmonitor filter
- yadif_cuda filter
version 4.0:
- Bitstream filters for editing metadata in H.264, HEVC and MPEG-2 streams
- Dropped support for OpenJPEG versions 2.0 and below. Using OpenJPEG now
requires 2.1 (or later) and pkg-config.
- VDA dropped (use VideoToolbox instead)
- MagicYUV encoder
- Raw AMR-NB and AMR-WB demuxers
- TiVo ty/ty+ demuxer
- Intel QSV-accelerated MJPEG encoding
- PCE support for extended channel layouts in the AAC encoder
- native aptX and aptX HD encoder and decoder
- Raw aptX and aptX HD muxer and demuxer
- NVIDIA NVDEC-accelerated H.264, HEVC, MJPEG, MPEG-1/2/4, VC1, VP8/9 hwaccel decoding
- Intel QSV-accelerated overlay filter
- mcompand audio filter
- acontrast audio filter
- OpenCL overlay filter
- video mix filter
- video normalize filter
- audio lv2 wrapper filter
- VAAPI MJPEG and VP8 decoding
- AMD AMF H.264 and HEVC encoders
- video fillborders filter
- video setrange filter
- nsp demuxer
- support LibreSSL (via libtls)
- AVX-512/ZMM support added
- Dropped support for building for Windows XP. The minimum supported Windows
version is Windows Vista.
- deconvolve video filter
- entropy video filter
- hilbert audio filter source
- aiir audio filter
- aiff: add support for CD-ROM XA ADPCM
- Removed the ffserver program
- Removed the ffmenc and ffmdec muxer and demuxer
- VideoToolbox HEVC encoder and hwaccel
- VAAPI-accelerated ProcAmp (color balance), denoise and sharpness filters
- Add android_camera indev
- codec2 en/decoding via libcodec2
- muxer/demuxer for raw codec2 files and .c2 files
- Moved nvidia codec headers into an external repository.
They can be found at http://git.videolan.org/?p=ffmpeg/nv-codec-headers.git
- native SBC encoder and decoder
- drmeter audio filter
- hapqa_extract bitstream filter
- filter_units bitstream filter
- AV1 Support through libaom
- E-AC-3 dependent frames support
- bitstream filter for extracting E-AC-3 core
- Haivision SRT protocol via libsrt
- segafilm muxer
- vfrdet filter
- SRCNN filter
version 3.4.1:
- avcodec/vp9_superframe_split_bsf: Fix integer overflow in frame_size/total_size checks
- avcodec/amrwbdec: Fix division by 0 in voice_factor()
- avformat/utils: Fix warning: ISO C90 forbids mixed declarations and code
- avcodec/decode: reset codec on receiving packet after EOF in compat_decode
- avcodec/diracdsp: Fix integer overflow in PUT_SIGNED_RECT_CLAMPED()
- avcodec/dirac_dwt: Fix integer overflows in COMPOSE_DAUB97*
- avcodec/extract_extradata_bsf: Fix leak discovered via fuzzing
- avcodec/vorbis: Fix another 1 << 31 > int32_t::max() with 1u.
- avcodec/vorbis: 1 << 31 > int32_t::max(), so use 1u << 31 instead.
- avformat/utils: Prevent undefined shift with wrap_bits > 64.
- avcodec/j2kenc: Fix out of array access in encode_cblk()
- avcodec/hevcdsp_template: Fix undefined shift in put_hevc_epel_bi_w_h()
- lavf/mov: fix huge alloc in mov_read_ctts
- avcodec/mlpdsp: Fix signed integer overflow, 2nd try
- avcodec/h264idct_template: Fix integer overflow in ff_h264_idct8_add
- avcodec/kgv1dec: Check that there is enough input for maximum RLE compression
- avformat/aacdec: Fix leak in adts_aac_read_packet()
- avcodec/dirac_dwt: Fix integer overflow in COMPOSE_FIDELITYi*
- avcodec/sbrdsp_fixed: Fix integer overflow
- avcodec/mpeg4videodec: Check also for negative versions in the validity check
- Close ogg stream upon error when using AV_EF_EXPLODE.
- Fix undefined shift on assumed 8-bit input.
- Use ff_thread_once for fixed, float table init.
- Fix leak of frame_duration_buffer in mov_fix_index().
- avformat/mov: Propagate errors in mov_switch_root.
- avcodec/hevcdsp_template: Fix invalid shift in put_hevc_epel_bi_w_v()
- avcodec/mlpdsp: Fix undefined shift ff_mlp_pack_output()
- avcodec/zmbv: Check that the buffer is large enough for mvec
- avcodec/dirac_dwt: Fix integer overflow in COMPOSE_DD137iL0()
- avcodec/wmv2dec: Check end of bitstream in parse_mb_skip() and ff_wmv2_decode_mb()
- avcodec/snowdec: Check for remaining bitstream in decode_blocks()
- avcodec/snowdec: Check intra block dc differences.
- avformat/mov: Check size of STSC allocation
- avcodec/vc2enc: Clear coef_buf on allocation
- avcodec/h264dec: Fix potential array overread
- avcodec/x86/mpegvideodsp: Fix signedness bug in need_emu
- avcodec/aacpsdsp_template: Fix integer overflows in ps_decorrelate_c()
- avcodec/aacdec_fixed: Fix undefined shift
- avcodec/mdct_*: Fix integer overflow in addition in RESCALE()
- avcodec/snowdec: Fix integer overflow in header parsing
- avcodec/cngdec: Fix integer clipping
- avcodec/sbrdsp_fixed: Fix integer overflow in shift in sbr_hf_g_filt_c()
- avcodec/aacsbr_fixed: Fix division by zero in sbr_gain_calc()
- avutil/softfloat: Add FLOAT_MIN
- avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
- avcodec/xan: Check for bitstream end in xan_huffman_decode()
- avcodec/exr: fix undefined shift in pxr24_uncompress()
- avformat: Free the internal codec context at the end
- avcodec/h264idct_template: Fix integer overflows in ff_h264_idct8_add()
- avcodec/xan: Improve overlapping check
- avcodec/aacdec_fixed: Fix integer overflow in apply_dependent_coupling_fixed()
- avcodec/aacdec_fixed: Fix integer overflow in predict()
- avcodec/jpeglsdec: Check for end of bitstream in ls_decode_line()
- avcodec/jpeglsdec: Check ilv for being a supported value
- tests/ffserver.regression.ref: update checksums to what ffserver currently produces
- ffserver: Fix off by 1 error in path
- avcodec/proresdec: align dequantization matrix buffers
- avformat/matroskaenc: add missing allocation failure checks for stream durations
- avformat/matroskaenc: actually enforce the stream limit
- configure: Fix dependencies of aac_at decoder.
- Don't manipulate duration when it's AV_NOPTS_VALUE.
- lavfi/af_pan: fix sign handling in channel coefficient parser
- avformat/hlsenc: write fmp4 init header after first AV frame
- avformat/hlsenc: allocate space for terminating null
- avformat/hlsenc: reindent hlsenc code
- avformat/hlsenc: check hls segment mode for ignore the init filename
- avformat/hlsenc: reindent hlsenc code
- avformat/hlsenc: fix missing first segment bug in fmp4 mode
- avformat/hlsenc: fix base_output_dirname is null when basename_size is 0 bug
- ffplay: use SDL2 audio API
- ffplay: only use hardware accelerated SDL texture formats
- ffplay: create the window and the renderer before starting playback
- ffmpeg: always init output stream before reaping filters
- vc2enc_dwt: pad the temporary buffer by the slice size
- lavu/arm: Check for have_vfp_vm instead of !have_vfpv3 for float_dsp_vfp
version 3.4:
- deflicker video filter

View File

@@ -1,4 +1,4 @@
## Installing FFmpeg
#Installing FFmpeg:
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.

View File

@@ -21,11 +21,10 @@ Specifically, the GPL parts of FFmpeg are:
- `compat/solaris/make_sunver.pl`
- `doc/t2h.pm`
- `doc/texi2pod.pl`
- `libswresample/tests/swresample.c`
- `libswresample/swresample-test.c`
- `tests/checkasm/*`
- `tests/tiny_ssim.c`
- the following filters in libavfilter:
- `signature_lookup.c`
- `vf_blackframe.c`
- `vf_boxblur.c`
- `vf_colormatrix.c`
@@ -35,13 +34,13 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_eq.c`
- `vf_find_rect.c`
- `vf_fspp.c`
- `vf_geq.c`
- `vf_histeq.c`
- `vf_hqdn3d.c`
- `vf_interlace.c`
- `vf_kerndeint.c`
- `vf_lensfun.c` (GPL version 3 or later)
- `vf_mcdeint.c`
- `vf_mpdecimate.c`
- `vf_nnedi.c`
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
@@ -50,14 +49,12 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_pullup.c`
- `vf_repeatfields.c`
- `vf_sab.c`
- `vf_signature.c`
- `vf_smartblur.c`
- `vf_spp.c`
- `vf_stereo3d.c`
- `vf_super2xsai.c`
- `vf_tinterlace.c`
- `vf_uspp.c`
- `vf_vaguedenoiser.c`
- `vsrc_mptestsrc.c`
Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -83,47 +80,41 @@ affect the licensing of binaries resulting from the combination.
### Compatible libraries
The following libraries are under GPL version 2:
- avisynth
The following libraries are under GPL:
- frei0r
- libcdio
- libdavs2
- librubberband
- libvidstab
- libx264
- libx265
- libxavs
- libxavs2
- libxvid
When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
passing `--enable-gpl` to configure.
The following libraries are under LGPL version 3:
- gmp
- libaribb24
- liblensfun
When combining them with FFmpeg, use the configure option `--enable-version3` to
upgrade FFmpeg to the LGPL v3.
The VMAF, mbedTLS, RK MPI, OpenCORE and VisualOn libraries are under the Apache License
2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with
The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
license is incompatible with the LGPL v2.1 and the GPL v2, but not with
version 3 of those licenses. So to combine these libraries with FFmpeg, the
license version needs to be upgraded by passing `--enable-version3` to configure.
The smbclient library is under the GPL v3, to combine it with FFmpeg,
the options `--enable-gpl` and `--enable-version3` have to be passed to
configure to upgrade FFmpeg to the GPL v3.
### Incompatible libraries
There are certain libraries you can combine with FFmpeg whose licenses are not
compatible with the GPL and/or the LGPL. If you wish to enable these
libraries, even in circumstances that their license may be incompatible, pass
`--enable-nonfree` to configure. This will cause the resulting binary to be
`--enable-nonfree` to configure. But note that if you enable any of these
libraries the resulting binary will be under a complex license mix that is
more restrictive than the LGPL and that may result in additional obligations.
It is possible that these restrictions cause the resulting binary to be
unredistributable.
The Fraunhofer FDK AAC and OpenSSL libraries are under licenses which are
incompatible with the GPLv2 and v3. To the best of our knowledge, they are
compatible with the LGPL.
The NVENC library, while its header file is licensed under the compatible MIT
license, requires a proprietary binary blob at run time, and is deemed to be
incompatible with the GPL. We are not certain if it is compatible with the
LGPL, but we require `--enable-nonfree` even with LGPL configurations in case
it is not.

View File

@@ -29,6 +29,9 @@ ffplay:
ffprobe:
ffprobe.c Stefano Sabatini
ffserver:
ffserver.c Reynaldo H. Verdejo Pinochet
Commandline utility code:
cmdutils.c, cmdutils.h Michael Niedermayer
@@ -39,7 +42,7 @@ QuickTime faststart:
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
project server Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov
presets Robert Swain
metadata subsystem Aurelien Jacobs
@@ -52,12 +55,12 @@ Communication
website Deby Barbara Lepage
fate.ffmpeg.org Timothy Gu
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
Patchwork Andriy Gelman
mailing lists Baptiste Coudurier
Twitter Reynaldo H. Verdejo Pinochet
Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
mailing lists Baptiste Coudurier, Lou Logan
Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser
Twitter Lou Logan, Reynaldo H. Verdejo Pinochet
Launchpad Timothy Gu
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, rcombs, wm4
ffmpeg-security Andreas Cadhalpun, Carl Eugen Hoyos, Clément Bœsch, Michael Niedermayer, Reimar Doeffinger, Rodger Combs, wm4
libavutil
@@ -78,7 +81,6 @@ Other:
float_dsp Loren Merritt
hash Reimar Doeffinger
hwcontext_cuda* Timo Rothenpieler
hwcontext_vulkan* Lynne
intfloat* Michael Niedermayer
integer.c, integer.h Michael Niedermayer
lzo Reimar Doeffinger
@@ -89,7 +91,6 @@ Other:
rational.c, rational.h Michael Niedermayer
rc4 Reimar Doeffinger
ripemd.c, ripemd.h James Almer
tx* Lynne
libavcodec
@@ -123,6 +124,7 @@ Generic Parts:
motion* Michael Niedermayer
rate control:
ratecontrol.c Michael Niedermayer
libxvid_rc.c Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h Michael Niedermayer
postprocessing:
@@ -140,12 +142,10 @@ Codecs:
aacenc*, aaccoder.c Rostislav Pehlivanov
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
aptx.c Aurelien Jacobs
ass* Aurelien Jacobs
asv* Michael Niedermayer
atrac3plus* Maxim Poliakovski
audiotoolbox* rcombs
avs2* Huiwen Ren
audiotoolbox* Rodger Combs
bgmc.c, bgmc.h Thilo Borgmann
binkaudio.c Peter Ross
cavs* Stefan Gehrer
@@ -158,7 +158,7 @@ Codecs:
cpia.c Stephan Hilb
crystalhd.c Philip Langdale
cscd.c Reimar Doeffinger
cuviddec.c Timo Rothenpieler
cuvid.c Timo Rothenpieler
dca* foo86
dirac* Rostislav Pehlivanov
dnxhd* Baptiste Coudurier
@@ -189,19 +189,15 @@ Codecs:
jvdec.c Peter Ross
lcl*.c Roberto Togni, Reimar Doeffinger
libcelt_dec.c Nicolas George
libcodec2.c Tomas Härdin
libdirac* David Conrad
libdavs2.c Huiwen Ren
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libvpx* James Zern
libxavs.c Stefan Gehrer
libxavs2.c Huiwen Ren
libzvbi-teletextdec.c Marton Balint
lzo.h, lzo.c Reimar Doeffinger
mdec.c Michael Niedermayer
@@ -216,8 +212,7 @@ Codecs:
msrle.c Mike Melanson
msvideo1.c Mike Melanson
nuv.c Reimar Doeffinger
nvdec*, nvenc* Timo Rothenpieler
omx.c Martin Storsjo, Aman Gupta
nvenc* Timo Rothenpieler
opus* Rostislav Pehlivanov
paf.* Paul B Mahol
pcx.c Ivo van Poorten
@@ -225,7 +220,7 @@ Codecs:
ptx.c Ivo van Poorten
qcelp* Reynaldo H. Verdejo Pinochet
qdm2.c, qdm2data.h Roberto Togni
qsv* Mark Thompson, Zhong Li
qsv* Mark Thompson
qtrle.c Mike Melanson
ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni
resample2.c Michael Niedermayer
@@ -235,6 +230,7 @@ Codecs:
rv10.c Michael Niedermayer
s3tc* Ivo van Poorten
smc.c Mike Melanson
smvjpegdec.c Ash Hughes
snow* Michael Niedermayer, Loren Merritt
sonic.c Alex Beregszaszi
speedhq.c Steinar H. Gunderson
@@ -246,10 +242,10 @@ Codecs:
tta.c Alex Beregszaszi, Jaikrishnan Menon
ttaenc.c Paul B Mahol
txd.c Ivo van Poorten
v4l2_* Jorge Ramirez-Ortiz
vc2* Rostislav Pehlivanov
vcr1.c Michael Niedermayer
videotoolboxenc.c Rick Kern, Aman Gupta
vda_h264_dec.c Xidorn Quan
videotoolboxenc.c Rick Kern
vima.c Paul B Mahol
vorbisdec.c Denes Balatoni, David Conrad
vorbisenc.c Oded Shimon
@@ -272,11 +268,11 @@ Hardware acceleration:
crystalhd.c Philip Langdale
dxva2* Hendrik Leppkes, Laurent Aimar, Steve Lhomme
d3d11va* Steve Lhomme
mediacodec* Matthieu Bouron, Aman Gupta
mediacodec* Matthieu Bouron
vaapi* Gwenole Beauchesne
vaapi_encode* Mark Thompson
vdpau* Philip Langdale, Carl Eugen Hoyos
videotoolbox* Rick Kern, Aman Gupta
videotoolbox* Rick Kern
libavdevice
@@ -286,7 +282,6 @@ libavdevice
avfoundation.m Thilo Borgmann
android_camera.c Felix Matouschek
decklink* Marton Balint
dshow.c Roger Pack (CC rogerdpack@gmail.com)
fbdev_enc.c Lukasz Marek
@@ -337,7 +332,6 @@ Filters:
vf_bwdif Thomas Mundt (CC <thomas.mundt@hr.de>)
vf_chromakey.c Timo Rothenpieler
vf_colorchannelmixer.c Paul B Mahol
vf_colorconstancy.c Mina Sami (CC <minas.gorgy@gmail.com>)
vf_colorbalance.c Paul B Mahol
vf_colorkey.c Timo Rothenpieler
vf_colorlevels.c Paul B Mahol
@@ -365,15 +359,12 @@ Filters:
vf_ssim.c Paul B Mahol
vf_stereo3d.c Paul B Mahol
vf_telecine.c Paul B Mahol
vf_tonemap_opencl.c Ruiling Song
vf_yadif.c Michael Niedermayer
vf_zoompan.c Paul B Mahol
Sources:
vsrc_mandelbrot.c Michael Niedermayer
dnn Yejun Guo
libavformat
===========
@@ -392,12 +383,7 @@ Muxers/Demuxers:
afc.c Paul B Mahol
aiffdec.c Baptiste Coudurier, Matthieu Bouron
aiffenc.c Baptiste Coudurier, Matthieu Bouron
alp.c Zane van Iperen
amvenc.c Zane van Iperen
apm.c Zane van Iperen
apngdec.c Benoit Fouet
argo_asf.c Zane van Iperen
argo_brp.c Zane van Iperen
ass* Aurelien Jacobs
astdec.c Paul B Mahol
astenc.c James Almer
@@ -409,10 +395,8 @@ Muxers/Demuxers:
brstm.c Paul B Mahol
caf* Peter Ross
cdxl.c Paul B Mahol
codec2.c Tomas Härdin
crc.c Michael Niedermayer
dashdec.c Steven Liu
dashenc.c Karthick Jeyapal
daud.c Reimar Doeffinger
dss.c Oleksij Rempel
dtsdec.c foo86
@@ -426,6 +410,7 @@ Muxers/Demuxers:
flvenc.c Michael Niedermayer, Steven Liu
gxf.c Reimar Doeffinger
gxfenc.c Baptiste Coudurier
hls.c Anssi Hannula
hlsenc.c Christian Suloway, Steven Liu
idcin.c Mike Melanson
idroqdec.c Mike Melanson
@@ -435,15 +420,14 @@ Muxers/Demuxers:
ircam* Paul B Mahol
iss.c Stefan Gehrer
jvdec.c Peter Ross
kvag.c Zane van Iperen
libmodplug.c Clément Bœsch
libopenmpt.c Josh de Kock
lmlm4.c Ivo van Poorten
lvfdec.c Paul B Mahol
lxfdec.c Tomas Härdin
matroska.c Aurelien Jacobs, Andreas Rheinhardt
matroskadec.c Aurelien Jacobs, Andreas Rheinhardt
matroskaenc.c David Conrad, Andreas Rheinhardt
matroska.c Aurelien Jacobs
matroskadec.c Aurelien Jacobs
matroskaenc.c David Conrad
matroska subtitles (matroskaenc.c) John Peebles
metadata* Aurelien Jacobs
mgsts.c Paul B Mahol
@@ -458,7 +442,7 @@ Muxers/Demuxers:
mpegtsenc.c Baptiste Coudurier
msnwc_tcp.c Ramiro Polla
mtv.c Reynaldo H. Verdejo Pinochet
mxf* Baptiste Coudurier, Tomas Härdin
mxf* Baptiste Coudurier
nistspheredec.c Paul B Mahol
nsvdec.c Francois Revol
nut* Michael Niedermayer
@@ -466,9 +450,9 @@ Muxers/Demuxers:
oggdec.c, oggdec.h David Conrad
oggenc.c Baptiste Coudurier
oggparse*.c David Conrad
oggparsedaala* Rostislav Pehlivanov
oma.c Maxim Poliakovski
paf.c Paul B Mahol
pp_bnk.c Zane van Iperen
psxstr.c Mike Melanson
pva.c Ivo van Poorten
pvfdec.c Paul B Mahol
@@ -514,7 +498,6 @@ Protocols:
ftp.c Lukasz Marek
http.c Ronald S. Bultje
libssh.c Lukasz Marek
libzmq.c Andriy Gelman
mms*.c Ronald S. Bultje
udp.c Luca Abeni
icecast.c Marvin Scholz
@@ -538,10 +521,9 @@ Operating systems / CPU architectures
=====================================
Alpha Falk Hueffner
MIPS Manojkumar Bhosale, Shiyou Yin
MIPS Manojkumar Bhosale
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC Lauri Kasanen
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
Windows MSVC Matthew Oliver, Hendrik Leppkes
@@ -567,10 +549,8 @@ Ivan Uskov
James Darnley
Jan Ekström
Joakim Plate
Jun Zhao
Kieran Kunhya
Kirill Gavrilov
Limin Wang
Martin Storsjö
Panagiotis Issaris
Pedro Arthur
@@ -591,11 +571,8 @@ Releases
If you want to maintain an older release, please contact us
GnuPG Fingerprints and IRC nicknames of maintainers and contributors
====================================================================
IRC nicknames are in parentheses. These apply
to the IRC channels listed on the website.
GnuPG Fingerprints of maintainers and contributors
==================================================
Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
@@ -610,17 +587,15 @@ FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
James Almer 7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
Lynne FE50 139C 6805 72CA FD52 1F8D A2FE A5F0 3F03 4464
Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
Nikolay Aleksandrov 8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
Philip Langdale 5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
Ramiro Polla 7859 C65B 751B 1179 792E DAE8 8E95 8B2F 9B6C 5700
Reimar Doeffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
@@ -629,9 +604,7 @@ Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
Thilo Borgmann (thilo) CE1D B7F4 4D20 FC3A DD9F FE5A 257C 5B8F 1D20 B92F
Tiancheng "Timothy" Gu 9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
Tim Nicholson 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
Tomas Härdin (thardin) A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
Zane van Iperen (zane) 61AE D40F 368B 6F26 9DAE 3892 6861 6B2D 8AC4 DCC5

View File

@@ -45,36 +45,20 @@ FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
tools/sofa2wavs$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
tools/target_dem_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
CONFIGURABLE_COMPONENTS = \
$(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) \
$(SRC_PATH)/libavcodec/bitstream_filters.c \
$(SRC_PATH)/libavcodec/parsers.c \
$(SRC_PATH)/libavformat/protocols.c \
config.h: ffbuild/.config
@@ -110,7 +94,7 @@ include $(SRC_PATH)/fftools/Makefile
include $(SRC_PATH)/doc/Makefile
include $(SRC_PATH)/doc/examples/Makefile
libavcodec/avcodec.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
ifeq ($(STRIPTYPE),direct)
@@ -144,25 +128,25 @@ install-data: $(DATA_FILES)
$(Q)mkdir -p "$(DATADIR)"
$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
uninstall: uninstall-data uninstall-headers uninstall-libs uninstall-pkgconfig
uninstall: uninstall-libs uninstall-headers uninstall-data
uninstall-data:
$(RM) -r "$(DATADIR)"
clean::
$(RM) $(CLEANSUFFIXES)
$(RM) $(addprefix compat/,$(CLEANSUFFIXES)) $(addprefix compat/*/,$(CLEANSUFFIXES)) $(addprefix compat/*/*/,$(CLEANSUFFIXES))
$(RM) $(CLEANSUFFIXES:%=compat/msvcrt/%)
$(RM) $(CLEANSUFFIXES:%=compat/atomics/pthread/%)
$(RM) $(CLEANSUFFIXES:%=compat/%)
$(RM) -r coverage-html
$(RM) -rf coverage.info coverage.info.in lcov
distclean:: clean
distclean::
$(RM) $(DISTCLEANSUFFIXES)
$(RM) .version avversion.h config.asm config.h mapfile \
ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
version.h libavutil/ffversion.h libavcodec/codec_names.h \
libavcodec/bsf_list.c libavformat/protocol_list.c \
libavcodec/codec_list.c libavcodec/parser_list.c \
libavfilter/filter_list.c libavdevice/indev_list.c libavdevice/outdev_list.c \
libavformat/muxer_list.c libavformat/demuxer_list.c
libavcodec/bsf_list.c libavformat/protocol_list.c
ifeq ($(SRC_LINK),src)
$(RM) src
endif
@@ -171,12 +155,11 @@ endif
config:
$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
build: all alltools examples testprogs
check: all alltools examples testprogs fate
include $(SRC_PATH)/tests/Makefile
$(sort $(OUTDIRS)):
$(sort $(OBJDIRS)):
$(Q)mkdir -p $@
# Dummy rule to stop make trying to rebuild removed or renamed headers
@@ -187,5 +170,4 @@ $(sort $(OUTDIRS)):
# so this saves some time on slow systems.
.SUFFIXES:
.PHONY: all all-yes alltools build check config testprogs
.PHONY: *clean install* uninstall*
.PHONY: all all-yes alltools check *clean config install* testprogs uninstall*

View File

@@ -21,6 +21,8 @@ such as audio, video, subtitles and related metadata.
* [ffplay](https://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
* [ffprobe](https://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
multimedia content.
* [ffserver](https://ffmpeg.org/ffserver.html) is a multimedia streaming server
for live broadcasts.
* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
## Documentation

View File

@@ -1 +1 @@
4.4.git
3.4.1

15
RELEASE_NOTES Normal file
View File

@@ -0,0 +1,15 @@
┌───────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 3.4 "Cantor" │
└───────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 3.4 "Cantor", about 6
months after the release of FFmpeg 3.3.
A complete Changelog is available at the root of the project, and the
complete Git history on http://source.ffmpeg.org.
We hope you will like this release as much as we enjoyed working on it, and
as usual, if you have any questions about it, or any FFmpeg related topic,
feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
on the mailing-lists.

1064
compat/avisynth/avisynth_c.h Normal file

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,62 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_CAPI_H
#define AVS_CAPI_H
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef BUILDING_AVSCORE
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#endif //AVS_CAPI_H

View File

@@ -0,0 +1,55 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_CONFIG_H
#define AVS_CONFIG_H
// Undefine this to get cdecl calling convention
#define AVSC_USE_STDCALL 1
// NOTE TO PLUGIN AUTHORS:
// Because FRAME_ALIGN can be substantially higher than the alignment
// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
// alignment. They should always request the exact alignment value they need.
// This is to make sure that plugins work over the widest range of AviSynth
// builds possible.
#define FRAME_ALIGN 32
#if defined(_M_AMD64) || defined(__x86_64)
# define X86_64
#elif defined(_M_IX86) || defined(__i386__)
# define X86_32
#else
# error Unsupported CPU architecture.
#endif
#endif //AVS_CONFIG_H

View File

@@ -0,0 +1,51 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef AVS_TYPES_H
#define AVS_TYPES_H
// Define all types necessary for interfacing with avisynth.dll
// Raster types used by VirtualDub & Avisynth
typedef unsigned int Pixel32;
typedef unsigned char BYTE;
// Audio Sample information
typedef float SFLOAT;
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
#endif //AVS_TYPES_H

View File

@@ -0,0 +1,728 @@
// Avisynth C Interface Version 0.20
// Copyright 2003 Kevin Atkinson
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program; if not, write to the Free Software
// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
// MA 02110-1301 USA, or visit
// http://www.gnu.org/copyleft/gpl.html .
//
// As a special exception, I give you permission to link to the
// Avisynth C interface with independent modules that communicate with
// the Avisynth C interface solely through the interfaces defined in
// avisynth_c.h, regardless of the license terms of these independent
// modules, and to copy and distribute the resulting combined work
// under terms of your choice, provided that every copy of the
// combined work is accompanied by a complete copy of the source code
// of the Avisynth C interface and Avisynth itself (with the version
// used to produce the combined work), being distributed under the
// terms of the GNU General Public License plus this exception. An
// independent module is a module which is not derived from or based
// on Avisynth C Interface, such as 3rd-party filters, import and
// export plugins, or graphical user interfaces.
#ifndef __AVXSYNTH_C__
#define __AVXSYNTH_C__
#include "windowsPorts/windows2linux.h"
#include <stdarg.h>
#ifdef __cplusplus
# define EXTERN_C extern "C"
#else
# define EXTERN_C
#endif
#define AVSC_USE_STDCALL 1
#ifndef AVSC_USE_STDCALL
# define AVSC_CC __cdecl
#else
# define AVSC_CC __stdcall
#endif
#define AVSC_INLINE static __inline
#ifdef AVISYNTH_C_EXPORTS
# define AVSC_EXPORT EXTERN_C
# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
#else
# define AVSC_EXPORT EXTERN_C __declspec(dllexport)
# ifndef AVSC_NO_DECLSPEC
# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
# else
# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
# endif
#endif
#ifdef __GNUC__
typedef long long int INT64;
#else
typedef __int64 INT64;
#endif
/////////////////////////////////////////////////////////////////////
//
// Constants
//
#ifndef __AVXSYNTH_H__
enum { AVISYNTH_INTERFACE_VERSION = 3 };
#endif
enum {AVS_SAMPLE_INT8 = 1<<0,
AVS_SAMPLE_INT16 = 1<<1,
AVS_SAMPLE_INT24 = 1<<2,
AVS_SAMPLE_INT32 = 1<<3,
AVS_SAMPLE_FLOAT = 1<<4};
enum {AVS_PLANAR_Y=1<<0,
AVS_PLANAR_U=1<<1,
AVS_PLANAR_V=1<<2,
AVS_PLANAR_ALIGNED=1<<3,
AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
// Colorspace properties.
enum {AVS_CS_BGR = 1<<28,
AVS_CS_YUV = 1<<29,
AVS_CS_INTERLEAVED = 1<<30,
AVS_CS_PLANAR = 1<<31};
// Specific colorformats
enum {
AVS_CS_UNKNOWN = 0,
AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar
AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar
AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR // same as above
};
enum {
AVS_IT_BFF = 1<<0,
AVS_IT_TFF = 1<<1,
AVS_IT_FIELDBASED = 1<<2};
enum {
AVS_FILTER_TYPE=1,
AVS_FILTER_INPUT_COLORSPACE=2,
AVS_FILTER_OUTPUT_TYPE=9,
AVS_FILTER_NAME=4,
AVS_FILTER_AUTHOR=5,
AVS_FILTER_VERSION=6,
AVS_FILTER_ARGS=7,
AVS_FILTER_ARGS_INFO=8,
AVS_FILTER_ARGS_DESCRIPTION=10,
AVS_FILTER_DESCRIPTION=11};
enum { //SUBTYPES
AVS_FILTER_TYPE_AUDIO=1,
AVS_FILTER_TYPE_VIDEO=2,
AVS_FILTER_OUTPUT_TYPE_SAME=3,
AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
enum {
AVS_CACHE_NOTHING=0,
AVS_CACHE_RANGE=1,
AVS_CACHE_ALL=2,
AVS_CACHE_AUDIO=3,
AVS_CACHE_AUDIO_NONE=4,
AVS_CACHE_AUDIO_AUTO=5
};
#define AVS_FRAME_ALIGN 16
typedef struct AVS_Clip AVS_Clip;
typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoInfo
//
// AVS_VideoInfo is layed out identicly to VideoInfo
typedef struct AVS_VideoInfo {
int width, height; // width=0 means no video
unsigned fps_numerator, fps_denominator;
int num_frames;
int pixel_type;
int audio_samples_per_second; // 0 means no audio
int sample_type;
INT64 num_audio_samples;
int nchannels;
// Imagetype properties
int image_type;
} AVS_VideoInfo;
// useful functions of the above
AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
{ return (p->width!=0); }
AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
{ return (p->audio_samples_per_second!=0); }
AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_BGR); }
AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
{ return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
{ return !!(p->pixel_type&AVS_CS_YUV ); }
AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
{ return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
{ return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
{ return ((p->pixel_type & c_space) == c_space); }
AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
{ return ((p->pixel_type & property)==property ); }
AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
{ return !!(p->pixel_type & AVS_CS_PLANAR); }
AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_FIELDBASED); }
AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
{ return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_BFF); }
AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
{ return !!(p->image_type & AVS_IT_TFF); }
AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
{
switch (p->pixel_type) {
case AVS_CS_BGR24: return 24;
case AVS_CS_BGR32: return 32;
case AVS_CS_YUY2: return 16;
case AVS_CS_YV12:
case AVS_CS_I420: return 12;
default: return 0;
}
}
AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
{ return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes
AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
{ return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images
AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
{ if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); }
AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
{ return p->audio_samples_per_second; }
AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
{
switch (p->sample_type) {
case AVS_SAMPLE_INT8: return sizeof(signed char);
case AVS_SAMPLE_INT16: return sizeof(signed short);
case AVS_SAMPLE_INT24: return 3;
case AVS_SAMPLE_INT32: return sizeof(signed int);
case AVS_SAMPLE_FLOAT: return sizeof(float);
default: return 0;
}
}
AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
{ return p->nchannels*avs_bytes_per_channel_sample(p);}
AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
{ return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
{ return bytes / avs_bytes_per_audio_sample(p); }
AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
{ return samples * avs_bytes_per_audio_sample(p); }
AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
{ return p->nchannels; }
AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
{ return p->sample_type;}
// useful mutator
AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
{ p->image_type|=property; }
AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
{ p->image_type&=~property; }
AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
{ if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
{
unsigned x=numerator, y=denominator;
while (y) { // find gcd
unsigned t = x%y; x = y; y = t;
}
p->fps_numerator = numerator/x;
p->fps_denominator = denominator/x;
}
AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
{
return (x->pixel_type == y->pixel_type)
|| (avs_is_yv12(x) && avs_is_yv12(y));
}
/////////////////////////////////////////////////////////////////////
//
// AVS_VideoFrame
//
// VideoFrameBuffer holds information about a memory block which is used
// for video data. For efficiency, instances of this class are not deleted
// when the refcount reaches zero; instead they're stored in a linked list
// to be reused. The instances are deleted when the corresponding AVS
// file is closed.
// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrameBuffer {
unsigned char * data;
int data_size;
// sequence_number is incremented every time the buffer is changed, so
// that stale views can tell they're no longer valid.
long sequence_number;
long refcount;
} AVS_VideoFrameBuffer;
// VideoFrame holds a "window" into a VideoFrameBuffer.
// AVS_VideoFrame is layed out identicly to IVideoFrame
// DO NOT USE THIS STRUCTURE DIRECTLY
typedef struct AVS_VideoFrame {
int refcount;
AVS_VideoFrameBuffer * vfb;
int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture.
} AVS_VideoFrame;
// Access functions for AVS_VideoFrame
AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
return p->pitch;}
AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
return p->pitch;}
AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
return p->row_size; }
AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
int r;
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->row_size>>1;
else return 0;
case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
if (p->pitchUV) {
r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
if (r < p->pitchUV)
return r;
return p->row_size>>1;
} else return 0;
case AVS_PLANAR_Y_ALIGNED:
r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
if (r <= p->pitch)
return r;
return p->row_size;
}
return p->row_size;
}
AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
return p->height;}
AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
switch (plane) {
case AVS_PLANAR_U: case AVS_PLANAR_V:
if (p->pitchUV) return p->height>>1;
return 0;
}
return p->height;}
AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
return p->vfb->data + p->offset;}
AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
{
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;}
}
AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
return (p->refcount == 1 && p->vfb->refcount == 1);}
AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p)
{
if (avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else
return 0;
}
AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
{
if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
++p->vfb->sequence_number;
return p->vfb->data + p->offset;
} else if (plane==AVS_PLANAR_Y) {
return 0;
} else {
switch (plane) {
case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
default: return p->vfb->data + p->offset;
}
}
}
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
// makes a shallow copy of a video frame
AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
{avs_release_video_frame(f);}
AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
{return avs_copy_video_frame(f);}
#endif
/////////////////////////////////////////////////////////////////////
//
// AVS_Value
//
// Treat AVS_Value as a fat pointer. That is use avs_copy_value
// and avs_release_value appropiaty as you would if AVS_Value was
// a pointer.
// To maintain source code compatibility with future versions of the
// avisynth_c API don't use the AVS_Value directly. Use the helper
// functions below.
// AVS_Value is layed out identicly to AVSValue
typedef struct AVS_Value AVS_Value;
struct AVS_Value {
short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
// for some function e'rror
short array_size;
union {
void * clip; // do not use directly, use avs_take_clip
char boolean;
int integer;
INT64 integer64; // match addition of __int64 to avxplugin.h
float floating_pt;
const char * string;
const AVS_Value * array;
} d;
};
// AVS_Value should be initilized with avs_void.
// Should also set to avs_void after the value is released
// with avs_copy_value. Consider it the equalvent of setting
// a pointer to NULL
static const AVS_Value avs_void = {'v'};
AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
AVSC_API(void, avs_release_value)(AVS_Value);
AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
#if defined __cplusplus
}
#endif // __cplusplus
AVSC_INLINE int avs_as_bool(AVS_Value v)
{ return v.d.boolean; }
AVSC_INLINE int avs_as_int(AVS_Value v)
{ return v.d.integer; }
AVSC_INLINE const char * avs_as_string(AVS_Value v)
{ return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
AVSC_INLINE double avs_as_float(AVS_Value v)
{ return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
AVSC_INLINE const char * avs_as_error(AVS_Value v)
{ return avs_is_error(v) ? v.d.string : 0; }
AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
{ return v.d.array; }
AVSC_INLINE int avs_array_size(AVS_Value v)
{ return avs_is_array(v) ? v.array_size : 1; }
AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
{ return avs_is_array(v) ? v.d.array[index] : v; }
// only use these functions on am AVS_Value that does not already have
// an active value. Remember, treat AVS_Value as a fat pointer.
AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
{ AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
AVSC_INLINE AVS_Value avs_new_value_int(int v0)
{ AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
{ AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
AVSC_INLINE AVS_Value avs_new_value_float(float v0)
{ AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
{ AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
{ AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
#endif
AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
{ AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
/////////////////////////////////////////////////////////////////////
//
// AVS_Clip
//
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_release_clip)(AVS_Clip *);
AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
AVSC_API(int, avs_get_version)(AVS_Clip *);
AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
// The returned video frame must be released with avs_release_video_frame
AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
// return field parity if field_based, else parity of first field in frame
AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
INT64 start, INT64 count);
// start and count are in samples
AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
int cachehints, size_t frame_range);
#if defined __cplusplus
}
#endif // __cplusplus
// This is the callback type used by avs_add_function
typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
(AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
typedef struct AVS_FilterInfo AVS_FilterInfo;
struct AVS_FilterInfo
{
// these members should not be modified outside of the AVS_ApplyFunc callback
AVS_Clip * child;
AVS_VideoInfo vi;
AVS_ScriptEnvironment * env;
AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
INT64 start, INT64 count);
int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
int frame_range);
void (AVSC_CC * free_filter)(AVS_FilterInfo *);
// Should be set when ever there is an error to report.
// It is cleared before any of the above methods are called
const char * error;
// this is to store whatever and may be modified at will
void * user_data;
};
// Create a new filter
// fi is set to point to the AVS_FilterInfo so that you can
// modify it once it is initilized.
// store_child should generally be set to true. If it is not
// set than ALL methods (the function pointers) must be defined
// If it is set than you do not need to worry about freeing the child
// clip.
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
AVS_FilterInfo * * fi,
AVS_Value child, int store_child);
#if defined __cplusplus
}
#endif // __cplusplus
/////////////////////////////////////////////////////////////////////
//
// AVS_ScriptEnvironment
//
// For GetCPUFlags. These are backwards-compatible with those in VirtualDub.
enum {
/* slowest CPU to support extension */
AVS_CPU_FORCE = 0x01, // N/A
AVS_CPU_FPU = 0x02, // 386/486DX
AVS_CPU_MMX = 0x04, // P55C, K6, PII
AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon
AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP
AVS_CPU_SSE2 = 0x20, // PIV, Hammer
AVS_CPU_3DNOW = 0x40, // K6-2
AVS_CPU_3DNOW_EXT = 0x80, // Athlon
AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2,
// which only Hammer will have anyway)
};
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
// note: val is really a va_list; I hope everyone typedefs va_list to a pointer
AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
const char * name, const char * params,
AVS_ApplyFunc apply, void * user_data);
AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
AVS_Value args, const char** arg_names);
// The returned value must be be released with avs_release_value
AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
// The returned value must be be released with avs_release_value
AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
//void avs_pop_context(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
const AVS_VideoInfo * vi, int align);
// align should be at least 16
#if defined __cplusplus
}
#endif // __cplusplus
#ifndef AVSC_NO_DECLSPEC
AVSC_INLINE
AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
AVSC_INLINE
AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
const AVS_VideoInfo * vi)
{return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
#endif
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
// The returned video frame must be be released
AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
// avisynth.dll exports this; it's a way to use it as a library, without
// writing an AVS script or without going through AVIFile.
AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
#if defined __cplusplus
}
#endif // __cplusplus
// this symbol is the entry point for the plugin and must
// be defined
AVSC_EXPORT
const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
#if defined __cplusplus
extern "C"
{
#endif // __cplusplus
AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
// The returned video frame must be be released
#if defined __cplusplus
}
#endif // __cplusplus
#endif //__AVXSYNTH_C__

View File

@@ -0,0 +1,85 @@
#ifndef __DATA_TYPE_CONVERSIONS_H__
#define __DATA_TYPE_CONVERSIONS_H__
#include <stdint.h>
#include <wchar.h>
#ifdef __cplusplus
namespace avxsynth {
#endif // __cplusplus
typedef int64_t __int64;
typedef int32_t __int32;
#ifdef __cplusplus
typedef bool BOOL;
#else
typedef uint32_t BOOL;
#endif // __cplusplus
typedef void* HMODULE;
typedef void* LPVOID;
typedef void* PVOID;
typedef PVOID HANDLE;
typedef HANDLE HWND;
typedef HANDLE HINSTANCE;
typedef void* HDC;
typedef void* HBITMAP;
typedef void* HICON;
typedef void* HFONT;
typedef void* HGDIOBJ;
typedef void* HBRUSH;
typedef void* HMMIO;
typedef void* HACMSTREAM;
typedef void* HACMDRIVER;
typedef void* HIC;
typedef void* HACMOBJ;
typedef HACMSTREAM* LPHACMSTREAM;
typedef void* HACMDRIVERID;
typedef void* LPHACMDRIVER;
typedef unsigned char BYTE;
typedef BYTE* LPBYTE;
typedef char TCHAR;
typedef TCHAR* LPTSTR;
typedef const TCHAR* LPCTSTR;
typedef char* LPSTR;
typedef LPSTR LPOLESTR;
typedef const char* LPCSTR;
typedef LPCSTR LPCOLESTR;
typedef wchar_t WCHAR;
typedef unsigned short WORD;
typedef unsigned int UINT;
typedef UINT MMRESULT;
typedef uint32_t DWORD;
typedef DWORD COLORREF;
typedef DWORD FOURCC;
typedef DWORD HRESULT;
typedef DWORD* LPDWORD;
typedef DWORD* DWORD_PTR;
typedef int32_t LONG;
typedef int32_t* LONG_PTR;
typedef LONG_PTR LRESULT;
typedef uint32_t ULONG;
typedef uint32_t* ULONG_PTR;
//typedef __int64_t intptr_t;
typedef uint64_t _fsize_t;
//
// Structures
//
typedef struct _GUID {
DWORD Data1;
WORD Data2;
WORD Data3;
BYTE Data4[8];
} GUID;
typedef GUID REFIID;
typedef GUID CLSID;
typedef CLSID* LPCLSID;
typedef GUID IID;
#ifdef __cplusplus
}; // namespace avxsynth
#endif // __cplusplus
#endif // __DATA_TYPE_CONVERSIONS_H__

View File

@@ -0,0 +1,77 @@
#ifndef __WINDOWS2LINUX_H__
#define __WINDOWS2LINUX_H__
/*
* LINUX SPECIFIC DEFINITIONS
*/
//
// Data types conversions
//
#include <stdlib.h>
#include <string.h>
#include "basicDataTypeConversions.h"
#ifdef __cplusplus
namespace avxsynth {
#endif // __cplusplus
//
// purposefully define the following MSFT definitions
// to mean nothing (as they do not mean anything on Linux)
//
#define __stdcall
#define __cdecl
#define noreturn
#define __declspec(x)
#define STDAPI extern "C" HRESULT
#define STDMETHODIMP HRESULT __stdcall
#define STDMETHODIMP_(x) x __stdcall
#define STDMETHOD(x) virtual HRESULT x
#define STDMETHOD_(a, x) virtual a x
#ifndef TRUE
#define TRUE true
#endif
#ifndef FALSE
#define FALSE false
#endif
#define S_OK (0x00000000)
#define S_FALSE (0x00000001)
#define E_NOINTERFACE (0X80004002)
#define E_POINTER (0x80004003)
#define E_FAIL (0x80004005)
#define E_OUTOFMEMORY (0x8007000E)
#define INVALID_HANDLE_VALUE ((HANDLE)((LONG_PTR)-1))
#define FAILED(hr) ((hr) & 0x80000000)
#define SUCCEEDED(hr) (!FAILED(hr))
//
// Functions
//
#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
#define MAKEWORD(a,b) (((a) << 8) | (b))
#define lstrlen strlen
#define lstrcpy strcpy
#define lstrcmpi strcasecmp
#define _stricmp strcasecmp
#define InterlockedIncrement(x) __sync_fetch_and_add((x), 1)
#define InterlockedDecrement(x) __sync_fetch_and_sub((x), 1)
// Windows uses (new, old) ordering but GCC has (old, new)
#define InterlockedCompareExchange(x,y,z) __sync_val_compare_and_swap(x,z,y)
#define UInt32x32To64(a, b) ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b)) ) )
#define Int64ShrlMod32(a, b) ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
#define Int32x32To64(a, b) ((__int64)(((__int64)((long)(a))) * ((long)(b))))
#define MulDiv(nNumber, nNumerator, nDenominator) (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
#ifdef __cplusplus
}; // namespace avxsynth
#endif // __cplusplus
#endif // __WINDOWS2LINUX_H__

View File

@@ -1,188 +0,0 @@
/*
* Minimum CUDA compatibility definitions header
*
* Copyright (c) 2019 rcombs
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_CUDA_CUDA_RUNTIME_H
#define COMPAT_CUDA_CUDA_RUNTIME_H
// Common macros
#define __global__ __attribute__((global))
#define __device__ __attribute__((device))
#define __device_builtin__ __attribute__((device_builtin))
#define __align__(N) __attribute__((aligned(N)))
#define __inline__ __inline__ __attribute__((always_inline))
#define max(a, b) ((a) > (b) ? (a) : (b))
#define min(a, b) ((a) < (b) ? (a) : (b))
#define abs(x) ((x) < 0 ? -(x) : (x))
#define atomicAdd(a, b) (__atomic_fetch_add(a, b, __ATOMIC_SEQ_CST))
// Basic typedefs
typedef __device_builtin__ unsigned long long cudaTextureObject_t;
typedef struct __device_builtin__ __align__(2) uchar2
{
unsigned char x, y;
} uchar2;
typedef struct __device_builtin__ __align__(4) ushort2
{
unsigned short x, y;
} ushort2;
typedef struct __device_builtin__ __align__(8) float2
{
float x, y;
} float2;
typedef struct __device_builtin__ __align__(8) int2
{
int x, y;
} int2;
typedef struct __device_builtin__ uint3
{
unsigned int x, y, z;
} uint3;
typedef struct uint3 dim3;
typedef struct __device_builtin__ __align__(4) uchar4
{
unsigned char x, y, z, w;
} uchar4;
typedef struct __device_builtin__ __align__(8) ushort4
{
unsigned short x, y, z, w;
} ushort4;
typedef struct __device_builtin__ __align__(16) int4
{
int x, y, z, w;
} int4;
typedef struct __device_builtin__ __align__(16) float4
{
float x, y, z, w;
} float4;
// Accessors for special registers
#define GETCOMP(reg, comp) \
asm("mov.u32 %0, %%" #reg "." #comp ";" : "=r"(tmp)); \
ret.comp = tmp;
#define GET(name, reg) static inline __device__ uint3 name() {\
uint3 ret; \
unsigned tmp; \
GETCOMP(reg, x) \
GETCOMP(reg, y) \
GETCOMP(reg, z) \
return ret; \
}
GET(getBlockIdx, ctaid)
GET(getBlockDim, ntid)
GET(getThreadIdx, tid)
// Instead of externs for these registers, we turn access to them into calls into trivial ASM
#define blockIdx (getBlockIdx())
#define blockDim (getBlockDim())
#define threadIdx (getThreadIdx())
// Basic initializers (simple macros rather than inline functions)
#define make_int2(a, b) ((int2){.x = a, .y = b})
#define make_uchar2(a, b) ((uchar2){.x = a, .y = b})
#define make_ushort2(a, b) ((ushort2){.x = a, .y = b})
#define make_float2(a, b) ((float2){.x = a, .y = b})
#define make_int4(a, b, c, d) ((int4){.x = a, .y = b, .z = c, .w = d})
#define make_uchar4(a, b, c, d) ((uchar4){.x = a, .y = b, .z = c, .w = d})
#define make_ushort4(a, b, c, d) ((ushort4){.x = a, .y = b, .z = c, .w = d})
#define make_float4(a, b, c, d) ((float4){.x = a, .y = b, .z = c, .w = d})
// Conversions from the tex instruction's 4-register output to various types
#define TEX2D(type, ret) static inline __device__ void conv(type* out, unsigned a, unsigned b, unsigned c, unsigned d) {*out = (ret);}
TEX2D(unsigned char, a & 0xFF)
TEX2D(unsigned short, a & 0xFFFF)
TEX2D(float, a)
TEX2D(uchar2, make_uchar2(a & 0xFF, b & 0xFF))
TEX2D(ushort2, make_ushort2(a & 0xFFFF, b & 0xFFFF))
TEX2D(float2, make_float2(a, b))
TEX2D(uchar4, make_uchar4(a & 0xFF, b & 0xFF, c & 0xFF, d & 0xFF))
TEX2D(ushort4, make_ushort4(a & 0xFFFF, b & 0xFFFF, c & 0xFFFF, d & 0xFFFF))
TEX2D(float4, make_float4(a, b, c, d))
// Template calling tex instruction and converting the output to the selected type
template<typename T>
inline __device__ T tex2D(cudaTextureObject_t texObject, float x, float y)
{
T ret;
unsigned ret1, ret2, ret3, ret4;
asm("tex.2d.v4.u32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
"=r"(ret1), "=r"(ret2), "=r"(ret3), "=r"(ret4) :
"l"(texObject), "f"(x), "f"(y));
conv(&ret, ret1, ret2, ret3, ret4);
return ret;
}
template<>
inline __device__ float4 tex2D<float4>(cudaTextureObject_t texObject, float x, float y)
{
float4 ret;
asm("tex.2d.v4.f32.f32 {%0, %1, %2, %3}, [%4, {%5, %6}];" :
"=r"(ret.x), "=r"(ret.y), "=r"(ret.z), "=r"(ret.w) :
"l"(texObject), "f"(x), "f"(y));
return ret;
}
template<>
inline __device__ float tex2D<float>(cudaTextureObject_t texObject, float x, float y)
{
return tex2D<float4>(texObject, x, y).x;
}
template<>
inline __device__ float2 tex2D<float2>(cudaTextureObject_t texObject, float x, float y)
{
float4 ret = tex2D<float4>(texObject, x, y);
return make_float2(ret.x, ret.y);
}
// Math helper functions
static inline __device__ float floorf(float a) { return __builtin_floorf(a); }
static inline __device__ float floor(float a) { return __builtin_floorf(a); }
static inline __device__ double floor(double a) { return __builtin_floor(a); }
static inline __device__ float ceilf(float a) { return __builtin_ceilf(a); }
static inline __device__ float ceil(float a) { return __builtin_ceilf(a); }
static inline __device__ double ceil(double a) { return __builtin_ceil(a); }
static inline __device__ float truncf(float a) { return __builtin_truncf(a); }
static inline __device__ float trunc(float a) { return __builtin_truncf(a); }
static inline __device__ double trunc(double a) { return __builtin_trunc(a); }
static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }
static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a); }
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */

View File

@@ -0,0 +1,98 @@
/*
* This copyright notice applies to this header file only:
*
* Copyright (c) 2016
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
#if !defined(AV_COMPAT_DYNLINK_CUDA_H) && !defined(CUDA_VERSION)
#define AV_COMPAT_DYNLINK_CUDA_H
#include <stddef.h>
#define CUDA_VERSION 7050
#if defined(_WIN32) || defined(__CYGWIN__)
#define CUDAAPI __stdcall
#else
#define CUDAAPI
#endif
#define CU_CTX_SCHED_BLOCKING_SYNC 4
typedef int CUdevice;
typedef void* CUarray;
typedef void* CUcontext;
typedef void* CUstream;
#if defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
typedef unsigned long long CUdeviceptr;
#else
typedef unsigned int CUdeviceptr;
#endif
typedef enum cudaError_enum {
CUDA_SUCCESS = 0
} CUresult;
typedef enum CUmemorytype_enum {
CU_MEMORYTYPE_HOST = 1,
CU_MEMORYTYPE_DEVICE = 2
} CUmemorytype;
typedef struct CUDA_MEMCPY2D_st {
size_t srcXInBytes;
size_t srcY;
CUmemorytype srcMemoryType;
const void *srcHost;
CUdeviceptr srcDevice;
CUarray srcArray;
size_t srcPitch;
size_t dstXInBytes;
size_t dstY;
CUmemorytype dstMemoryType;
void *dstHost;
CUdeviceptr dstDevice;
CUarray dstArray;
size_t dstPitch;
size_t WidthInBytes;
size_t Height;
} CUDA_MEMCPY2D;
typedef CUresult CUDAAPI tcuInit(unsigned int Flags);
typedef CUresult CUDAAPI tcuDeviceGetCount(int *count);
typedef CUresult CUDAAPI tcuDeviceGet(CUdevice *device, int ordinal);
typedef CUresult CUDAAPI tcuDeviceGetName(char *name, int len, CUdevice dev);
typedef CUresult CUDAAPI tcuDeviceComputeCapability(int *major, int *minor, CUdevice dev);
typedef CUresult CUDAAPI tcuCtxCreate_v2(CUcontext *pctx, unsigned int flags, CUdevice dev);
typedef CUresult CUDAAPI tcuCtxPushCurrent_v2(CUcontext *pctx);
typedef CUresult CUDAAPI tcuCtxPopCurrent_v2(CUcontext *pctx);
typedef CUresult CUDAAPI tcuCtxDestroy_v2(CUcontext ctx);
typedef CUresult CUDAAPI tcuMemAlloc_v2(CUdeviceptr *dptr, size_t bytesize);
typedef CUresult CUDAAPI tcuMemFree_v2(CUdeviceptr dptr);
typedef CUresult CUDAAPI tcuMemcpy2D_v2(const CUDA_MEMCPY2D *pcopy);
typedef CUresult CUDAAPI tcuGetErrorName(CUresult error, const char** pstr);
typedef CUresult CUDAAPI tcuGetErrorString(CUresult error, const char** pstr);
#endif

View File

@@ -0,0 +1,886 @@
/*
* This copyright notice applies to this header file only:
*
* Copyright (c) 2010-2017 NVIDIA Corporation
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
/*****************************************************************************************************/
//! \file cuviddec.h
//! NVDECODE API provides video decoding interface to NVIDIA GPU devices.
//! \date 2015-2017
//! This file contains constants, structure definitions and function prototypes used for decoding.
/*****************************************************************************************************/
#if !defined(__CUDA_VIDEO_H__)
#define __CUDA_VIDEO_H__
#if defined(_WIN64) || defined(__LP64__) || defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
#if (CUDA_VERSION >= 3020) && (!defined(CUDA_FORCE_API_VERSION) || (CUDA_FORCE_API_VERSION >= 3020))
#define __CUVID_DEVPTR64
#endif
#endif
#if defined(__cplusplus)
extern "C" {
#endif /* __cplusplus */
#if defined(__CYGWIN__)
typedef unsigned int tcu_ulong;
#else
typedef unsigned long tcu_ulong;
#endif
typedef void *CUvideodecoder;
typedef struct _CUcontextlock_st *CUvideoctxlock;
/*********************************************************************************/
//! \enum cudaVideoCodec
//! Video codec enums
//! These enums are used in CUVIDDECODECREATEINFO and CUVIDDECODECAPS structures
/*********************************************************************************/
typedef enum cudaVideoCodec_enum {
cudaVideoCodec_MPEG1=0, /**< MPEG1 */
cudaVideoCodec_MPEG2, /**< MPEG2 */
cudaVideoCodec_MPEG4, /**< MPEG4 */
cudaVideoCodec_VC1, /**< VC1 */
cudaVideoCodec_H264, /**< H264 */
cudaVideoCodec_JPEG, /**< JPEG */
cudaVideoCodec_H264_SVC, /**< H264-SVC */
cudaVideoCodec_H264_MVC, /**< H264-MVC */
cudaVideoCodec_HEVC, /**< HEVC */
cudaVideoCodec_VP8, /**< VP8 */
cudaVideoCodec_VP9, /**< VP9 */
cudaVideoCodec_NumCodecs, /**< Max codecs */
// Uncompressed YUV
cudaVideoCodec_YUV420 = (('I'<<24)|('Y'<<16)|('U'<<8)|('V')), /**< Y,U,V (4:2:0) */
cudaVideoCodec_YV12 = (('Y'<<24)|('V'<<16)|('1'<<8)|('2')), /**< Y,V,U (4:2:0) */
cudaVideoCodec_NV12 = (('N'<<24)|('V'<<16)|('1'<<8)|('2')), /**< Y,UV (4:2:0) */
cudaVideoCodec_YUYV = (('Y'<<24)|('U'<<16)|('Y'<<8)|('V')), /**< YUYV/YUY2 (4:2:2) */
cudaVideoCodec_UYVY = (('U'<<24)|('Y'<<16)|('V'<<8)|('Y')) /**< UYVY (4:2:2) */
} cudaVideoCodec;
/*********************************************************************************/
//! \enum cudaVideoSurfaceFormat
//! Video surface format enums used for output format of decoded output
//! These enums are used in CUVIDDECODECREATEINFO structure
/*********************************************************************************/
typedef enum cudaVideoSurfaceFormat_enum {
cudaVideoSurfaceFormat_NV12=0, /**< NV12 format */
cudaVideoSurfaceFormat_P016=1 /**< 16 bit semiplaner format. Can be used for 10 bit(6LSB bits 0),
12 bit (4LSB bits 0) */
} cudaVideoSurfaceFormat;
/******************************************************************************************************************/
//! \enum cudaVideoDeinterlaceMode
//! Deinterlacing mode enums
//! These enums are used in CUVIDDECODECREATEINFO structure
//! Use cudaVideoDeinterlaceMode_Weave for progressive content and for content that doesn't need deinterlacing
//! cudaVideoDeinterlaceMode_Adaptive needs more video memory than other DImodes
/******************************************************************************************************************/
typedef enum cudaVideoDeinterlaceMode_enum {
cudaVideoDeinterlaceMode_Weave=0, /**< Weave both fields (no deinterlacing) */
cudaVideoDeinterlaceMode_Bob, /**< Drop one field */
cudaVideoDeinterlaceMode_Adaptive /**< Adaptive deinterlacing */
} cudaVideoDeinterlaceMode;
/**************************************************************************************************************/
//! \enum cudaVideoChromaFormat
//! Chroma format enums
//! These enums are used in CUVIDDECODECREATEINFO and CUVIDDECODECAPS structures
//! JPEG supports Monochrome, YUV 4:2:0, YUV 4:2:2 and YUV 4:4:4 chroma formats.
//! H264, HEVC, VP9, VP8, VC1, MPEG1, MPEG2 and MPEG4 support YUV 4:2:0 chroma format only.
/**************************************************************************************************************/
typedef enum cudaVideoChromaFormat_enum {
cudaVideoChromaFormat_Monochrome=0, /**< MonoChrome */
cudaVideoChromaFormat_420, /**< YUV 4:2:0 */
cudaVideoChromaFormat_422, /**< YUV 4:2:2 */
cudaVideoChromaFormat_444 /**< YUV 4:4:4 */
} cudaVideoChromaFormat;
/*************************************************************************************************************/
//! \enum cudaVideoCreateFlags
//! Decoder flag enums to select preferred decode path
//! cudaVideoCreate_Default and cudaVideoCreate_PreferCUVID are most optimized, use these whenever possible
/*************************************************************************************************************/
typedef enum cudaVideoCreateFlags_enum {
cudaVideoCreate_Default = 0x00, /**< Default operation mode: use dedicated video engines */
cudaVideoCreate_PreferCUDA = 0x01, /**< Use CUDA-based decoder (requires valid vidLock object for multi-threading) */
cudaVideoCreate_PreferDXVA = 0x02, /**< Go through DXVA internally if possible (requires D3D9 interop) */
cudaVideoCreate_PreferCUVID = 0x04 /**< Use dedicated video engines directly */
} cudaVideoCreateFlags;
/**************************************************************************************************************/
//! \struct CUVIDDECODECAPS;
//! This structure is used in cuvidGetDecoderCaps API
/**************************************************************************************************************/
typedef struct _CUVIDDECODECAPS
{
cudaVideoCodec eCodecType; /**< IN: cudaVideoCodec_XXX */
cudaVideoChromaFormat eChromaFormat; /**< IN: cudaVideoChromaFormat_XXX */
unsigned int nBitDepthMinus8; /**< IN: The Value "BitDepth minus 8" */
unsigned int reserved1[3]; /**< Reserved for future use - set to zero */
unsigned char bIsSupported; /**< OUT: 1 if codec supported, 0 if not supported */
unsigned char reserved2[3]; /**< Reserved for future use - set to zero */
unsigned int nMaxWidth; /**< OUT: Max supported coded width in pixels */
unsigned int nMaxHeight; /**< OUT: Max supported coded height in pixels */
unsigned int nMaxMBCount; /**< OUT: Max supported macroblock count
CodedWidth*CodedHeight/256 must be <= nMaxMBCount */
unsigned short nMinWidth; /**< OUT: Min supported coded width in pixels */
unsigned short nMinHeight; /**< OUT: Min supported coded height in pixels */
unsigned int reserved3[11]; /**< Reserved for future use - set to zero */
} CUVIDDECODECAPS;
/**************************************************************************************************************/
//! \struct CUVIDDECODECREATEINFO
//! This structure is used in cuvidCreateDecoder API
/**************************************************************************************************************/
typedef struct _CUVIDDECODECREATEINFO
{
tcu_ulong ulWidth; /**< IN: Coded sequence width in pixels */
tcu_ulong ulHeight; /**< IN: Coded sequence height in pixels */
tcu_ulong ulNumDecodeSurfaces; /**< IN: Maximum number of internal decode surfaces */
cudaVideoCodec CodecType; /**< IN: cudaVideoCodec_XXX */
cudaVideoChromaFormat ChromaFormat; /**< IN: cudaVideoChromaFormat_XXX */
tcu_ulong ulCreationFlags; /**< IN: Decoder creation flags (cudaVideoCreateFlags_XXX) */
tcu_ulong bitDepthMinus8; /**< IN: The value "BitDepth minus 8" */
tcu_ulong ulIntraDecodeOnly; /**< IN: Set 1 only if video has all intra frames (default value is 0). This will
optimize video memory for Intra frames only decoding. The support is limited
to specific codecs(H264 rightnow), the flag will be ignored for codecs which
are not supported. However decoding might fail if the flag is enabled in case
of supported codecs for regular bit streams having P and/or B frames. */
tcu_ulong Reserved1[3]; /**< Reserved for future use - set to zero */
/**
* IN: area of the frame that should be displayed
*/
struct {
short left;
short top;
short right;
short bottom;
} display_area;
cudaVideoSurfaceFormat OutputFormat; /**< IN: cudaVideoSurfaceFormat_XXX */
cudaVideoDeinterlaceMode DeinterlaceMode; /**< IN: cudaVideoDeinterlaceMode_XXX */
tcu_ulong ulTargetWidth; /**< IN: Post-processed output width (Should be aligned to 2) */
tcu_ulong ulTargetHeight; /**< IN: Post-processed output height (Should be aligbed to 2) */
tcu_ulong ulNumOutputSurfaces; /**< IN: Maximum number of output surfaces simultaneously mapped */
CUvideoctxlock vidLock; /**< IN: If non-NULL, context lock used for synchronizing ownership of
the cuda context. Needed for cudaVideoCreate_PreferCUDA decode */
/**
* IN: target rectangle in the output frame (for aspect ratio conversion)
* if a null rectangle is specified, {0,0,ulTargetWidth,ulTargetHeight} will be used
*/
struct {
short left;
short top;
short right;
short bottom;
} target_rect;
tcu_ulong Reserved2[5]; /**< Reserved for future use - set to zero */
} CUVIDDECODECREATEINFO;
/*********************************************************/
//! \struct CUVIDH264DPBENTRY
//! H.264 DPB entry
//! This structure is used in CUVIDH264PICPARAMS structure
/*********************************************************/
typedef struct _CUVIDH264DPBENTRY
{
int PicIdx; /**< picture index of reference frame */
int FrameIdx; /**< frame_num(short-term) or LongTermFrameIdx(long-term) */
int is_long_term; /**< 0=short term reference, 1=long term reference */
int not_existing; /**< non-existing reference frame (corresponding PicIdx should be set to -1) */
int used_for_reference; /**< 0=unused, 1=top_field, 2=bottom_field, 3=both_fields */
int FieldOrderCnt[2]; /**< field order count of top and bottom fields */
} CUVIDH264DPBENTRY;
/************************************************************/
//! \struct CUVIDH264MVCEXT
//! H.264 MVC picture parameters ext
//! This structure is used in CUVIDH264PICPARAMS structure
/************************************************************/
typedef struct _CUVIDH264MVCEXT
{
int num_views_minus1; /**< Max number of coded views minus 1 in video : Range - 0 to 1023 */
int view_id; /**< view identifier */
unsigned char inter_view_flag; /**< 1 if used for inter-view prediction, 0 if not */
unsigned char num_inter_view_refs_l0; /**< number of inter-view ref pics in RefPicList0 */
unsigned char num_inter_view_refs_l1; /**< number of inter-view ref pics in RefPicList1 */
unsigned char MVCReserved8Bits; /**< Reserved bits */
int InterViewRefsL0[16]; /**< view id of the i-th view component for inter-view prediction in RefPicList0 */
int InterViewRefsL1[16]; /**< view id of the i-th view component for inter-view prediction in RefPicList1 */
} CUVIDH264MVCEXT;
/*********************************************************/
//! \struct CUVIDH264SVCEXT
//! H.264 SVC picture parameters ext
//! This structure is used in CUVIDH264PICPARAMS structure
/*********************************************************/
typedef struct _CUVIDH264SVCEXT
{
unsigned char profile_idc;
unsigned char level_idc;
unsigned char DQId;
unsigned char DQIdMax;
unsigned char disable_inter_layer_deblocking_filter_idc;
unsigned char ref_layer_chroma_phase_y_plus1;
signed char inter_layer_slice_alpha_c0_offset_div2;
signed char inter_layer_slice_beta_offset_div2;
unsigned short DPBEntryValidFlag;
unsigned char inter_layer_deblocking_filter_control_present_flag;
unsigned char extended_spatial_scalability_idc;
unsigned char adaptive_tcoeff_level_prediction_flag;
unsigned char slice_header_restriction_flag;
unsigned char chroma_phase_x_plus1_flag;
unsigned char chroma_phase_y_plus1;
unsigned char tcoeff_level_prediction_flag;
unsigned char constrained_intra_resampling_flag;
unsigned char ref_layer_chroma_phase_x_plus1_flag;
unsigned char store_ref_base_pic_flag;
unsigned char Reserved8BitsA;
unsigned char Reserved8BitsB;
short scaled_ref_layer_left_offset;
short scaled_ref_layer_top_offset;
short scaled_ref_layer_right_offset;
short scaled_ref_layer_bottom_offset;
unsigned short Reserved16Bits;
struct _CUVIDPICPARAMS *pNextLayer; /**< Points to the picparams for the next layer to be decoded.
Linked list ends at the target layer. */
int bRefBaseLayer; /**< whether to store ref base pic */
} CUVIDH264SVCEXT;
/******************************************************/
//! \struct CUVIDH264PICPARAMS
//! H.264 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/******************************************************/
typedef struct _CUVIDH264PICPARAMS
{
// SPS
int log2_max_frame_num_minus4;
int pic_order_cnt_type;
int log2_max_pic_order_cnt_lsb_minus4;
int delta_pic_order_always_zero_flag;
int frame_mbs_only_flag;
int direct_8x8_inference_flag;
int num_ref_frames; // NOTE: shall meet level 4.1 restrictions
unsigned char residual_colour_transform_flag;
unsigned char bit_depth_luma_minus8; // Must be 0 (only 8-bit supported)
unsigned char bit_depth_chroma_minus8; // Must be 0 (only 8-bit supported)
unsigned char qpprime_y_zero_transform_bypass_flag;
// PPS
int entropy_coding_mode_flag;
int pic_order_present_flag;
int num_ref_idx_l0_active_minus1;
int num_ref_idx_l1_active_minus1;
int weighted_pred_flag;
int weighted_bipred_idc;
int pic_init_qp_minus26;
int deblocking_filter_control_present_flag;
int redundant_pic_cnt_present_flag;
int transform_8x8_mode_flag;
int MbaffFrameFlag;
int constrained_intra_pred_flag;
int chroma_qp_index_offset;
int second_chroma_qp_index_offset;
int ref_pic_flag;
int frame_num;
int CurrFieldOrderCnt[2];
// DPB
CUVIDH264DPBENTRY dpb[16]; // List of reference frames within the DPB
// Quantization Matrices (raster-order)
unsigned char WeightScale4x4[6][16];
unsigned char WeightScale8x8[2][64];
// FMO/ASO
unsigned char fmo_aso_enable;
unsigned char num_slice_groups_minus1;
unsigned char slice_group_map_type;
signed char pic_init_qs_minus26;
unsigned int slice_group_change_rate_minus1;
union
{
unsigned long long slice_group_map_addr;
const unsigned char *pMb2SliceGroupMap;
} fmo;
unsigned int Reserved[12];
// SVC/MVC
union
{
CUVIDH264MVCEXT mvcext;
CUVIDH264SVCEXT svcext;
};
} CUVIDH264PICPARAMS;
/********************************************************/
//! \struct CUVIDMPEG2PICPARAMS
//! MPEG-2 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/********************************************************/
typedef struct _CUVIDMPEG2PICPARAMS
{
int ForwardRefIdx; // Picture index of forward reference (P/B-frames)
int BackwardRefIdx; // Picture index of backward reference (B-frames)
int picture_coding_type;
int full_pel_forward_vector;
int full_pel_backward_vector;
int f_code[2][2];
int intra_dc_precision;
int frame_pred_frame_dct;
int concealment_motion_vectors;
int q_scale_type;
int intra_vlc_format;
int alternate_scan;
int top_field_first;
// Quantization matrices (raster order)
unsigned char QuantMatrixIntra[64];
unsigned char QuantMatrixInter[64];
} CUVIDMPEG2PICPARAMS;
// MPEG-4 has VOP types instead of Picture types
#define I_VOP 0
#define P_VOP 1
#define B_VOP 2
#define S_VOP 3
/*******************************************************/
//! \struct CUVIDMPEG4PICPARAMS
//! MPEG-4 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/*******************************************************/
typedef struct _CUVIDMPEG4PICPARAMS
{
int ForwardRefIdx; // Picture index of forward reference (P/B-frames)
int BackwardRefIdx; // Picture index of backward reference (B-frames)
// VOL
int video_object_layer_width;
int video_object_layer_height;
int vop_time_increment_bitcount;
int top_field_first;
int resync_marker_disable;
int quant_type;
int quarter_sample;
int short_video_header;
int divx_flags;
// VOP
int vop_coding_type;
int vop_coded;
int vop_rounding_type;
int alternate_vertical_scan_flag;
int interlaced;
int vop_fcode_forward;
int vop_fcode_backward;
int trd[2];
int trb[2];
// Quantization matrices (raster order)
unsigned char QuantMatrixIntra[64];
unsigned char QuantMatrixInter[64];
int gmc_enabled;
} CUVIDMPEG4PICPARAMS;
/********************************************************/
//! \struct CUVIDVC1PICPARAMS
//! VC1 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/********************************************************/
typedef struct _CUVIDVC1PICPARAMS
{
int ForwardRefIdx; /**< Picture index of forward reference (P/B-frames) */
int BackwardRefIdx; /**< Picture index of backward reference (B-frames) */
int FrameWidth; /**< Actual frame width */
int FrameHeight; /**< Actual frame height */
// PICTURE
int intra_pic_flag; /**< Set to 1 for I,BI frames */
int ref_pic_flag; /**< Set to 1 for I,P frames */
int progressive_fcm; /**< Progressive frame */
// SEQUENCE
int profile;
int postprocflag;
int pulldown;
int interlace;
int tfcntrflag;
int finterpflag;
int psf;
int multires;
int syncmarker;
int rangered;
int maxbframes;
// ENTRYPOINT
int panscan_flag;
int refdist_flag;
int extended_mv;
int dquant;
int vstransform;
int loopfilter;
int fastuvmc;
int overlap;
int quantizer;
int extended_dmv;
int range_mapy_flag;
int range_mapy;
int range_mapuv_flag;
int range_mapuv;
int rangeredfrm; // range reduction state
} CUVIDVC1PICPARAMS;
/***********************************************************/
//! \struct CUVIDJPEGPICPARAMS
//! JPEG picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/***********************************************************/
typedef struct _CUVIDJPEGPICPARAMS
{
int Reserved;
} CUVIDJPEGPICPARAMS;
/*******************************************************/
//! \struct CUVIDHEVCPICPARAMS
//! HEVC picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/*******************************************************/
typedef struct _CUVIDHEVCPICPARAMS
{
// sps
int pic_width_in_luma_samples;
int pic_height_in_luma_samples;
unsigned char log2_min_luma_coding_block_size_minus3;
unsigned char log2_diff_max_min_luma_coding_block_size;
unsigned char log2_min_transform_block_size_minus2;
unsigned char log2_diff_max_min_transform_block_size;
unsigned char pcm_enabled_flag;
unsigned char log2_min_pcm_luma_coding_block_size_minus3;
unsigned char log2_diff_max_min_pcm_luma_coding_block_size;
unsigned char pcm_sample_bit_depth_luma_minus1;
unsigned char pcm_sample_bit_depth_chroma_minus1;
unsigned char pcm_loop_filter_disabled_flag;
unsigned char strong_intra_smoothing_enabled_flag;
unsigned char max_transform_hierarchy_depth_intra;
unsigned char max_transform_hierarchy_depth_inter;
unsigned char amp_enabled_flag;
unsigned char separate_colour_plane_flag;
unsigned char log2_max_pic_order_cnt_lsb_minus4;
unsigned char num_short_term_ref_pic_sets;
unsigned char long_term_ref_pics_present_flag;
unsigned char num_long_term_ref_pics_sps;
unsigned char sps_temporal_mvp_enabled_flag;
unsigned char sample_adaptive_offset_enabled_flag;
unsigned char scaling_list_enable_flag;
unsigned char IrapPicFlag;
unsigned char IdrPicFlag;
unsigned char bit_depth_luma_minus8;
unsigned char bit_depth_chroma_minus8;
unsigned char reserved1[14];
// pps
unsigned char dependent_slice_segments_enabled_flag;
unsigned char slice_segment_header_extension_present_flag;
unsigned char sign_data_hiding_enabled_flag;
unsigned char cu_qp_delta_enabled_flag;
unsigned char diff_cu_qp_delta_depth;
signed char init_qp_minus26;
signed char pps_cb_qp_offset;
signed char pps_cr_qp_offset;
unsigned char constrained_intra_pred_flag;
unsigned char weighted_pred_flag;
unsigned char weighted_bipred_flag;
unsigned char transform_skip_enabled_flag;
unsigned char transquant_bypass_enabled_flag;
unsigned char entropy_coding_sync_enabled_flag;
unsigned char log2_parallel_merge_level_minus2;
unsigned char num_extra_slice_header_bits;
unsigned char loop_filter_across_tiles_enabled_flag;
unsigned char loop_filter_across_slices_enabled_flag;
unsigned char output_flag_present_flag;
unsigned char num_ref_idx_l0_default_active_minus1;
unsigned char num_ref_idx_l1_default_active_minus1;
unsigned char lists_modification_present_flag;
unsigned char cabac_init_present_flag;
unsigned char pps_slice_chroma_qp_offsets_present_flag;
unsigned char deblocking_filter_override_enabled_flag;
unsigned char pps_deblocking_filter_disabled_flag;
signed char pps_beta_offset_div2;
signed char pps_tc_offset_div2;
unsigned char tiles_enabled_flag;
unsigned char uniform_spacing_flag;
unsigned char num_tile_columns_minus1;
unsigned char num_tile_rows_minus1;
unsigned short column_width_minus1[21];
unsigned short row_height_minus1[21];
unsigned int reserved3[15];
// RefPicSets
int NumBitsForShortTermRPSInSlice;
int NumDeltaPocsOfRefRpsIdx;
int NumPocTotalCurr;
int NumPocStCurrBefore;
int NumPocStCurrAfter;
int NumPocLtCurr;
int CurrPicOrderCntVal;
int RefPicIdx[16]; // [refpic] Indices of valid reference pictures (-1 if unused for reference)
int PicOrderCntVal[16]; // [refpic]
unsigned char IsLongTerm[16]; // [refpic] 0=not a long-term reference, 1=long-term reference
unsigned char RefPicSetStCurrBefore[8]; // [0..NumPocStCurrBefore-1] -> refpic (0..15)
unsigned char RefPicSetStCurrAfter[8]; // [0..NumPocStCurrAfter-1] -> refpic (0..15)
unsigned char RefPicSetLtCurr[8]; // [0..NumPocLtCurr-1] -> refpic (0..15)
unsigned char RefPicSetInterLayer0[8];
unsigned char RefPicSetInterLayer1[8];
unsigned int reserved4[12];
// scaling lists (diag order)
unsigned char ScalingList4x4[6][16]; // [matrixId][i]
unsigned char ScalingList8x8[6][64]; // [matrixId][i]
unsigned char ScalingList16x16[6][64]; // [matrixId][i]
unsigned char ScalingList32x32[2][64]; // [matrixId][i]
unsigned char ScalingListDCCoeff16x16[6]; // [matrixId]
unsigned char ScalingListDCCoeff32x32[2]; // [matrixId]
} CUVIDHEVCPICPARAMS;
/***********************************************************/
//! \struct CUVIDVP8PICPARAMS
//! VP8 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/***********************************************************/
typedef struct _CUVIDVP8PICPARAMS
{
int width;
int height;
unsigned int first_partition_size;
//Frame Indexes
unsigned char LastRefIdx;
unsigned char GoldenRefIdx;
unsigned char AltRefIdx;
union {
struct {
unsigned char frame_type : 1; /**< 0 = KEYFRAME, 1 = INTERFRAME */
unsigned char version : 3;
unsigned char show_frame : 1;
unsigned char update_mb_segmentation_data : 1; /**< Must be 0 if segmentation is not enabled */
unsigned char Reserved2Bits : 2;
};
unsigned char wFrameTagFlags;
};
unsigned char Reserved1[4];
unsigned int Reserved2[3];
} CUVIDVP8PICPARAMS;
/***********************************************************/
//! \struct CUVIDVP9PICPARAMS
//! VP9 picture parameters
//! This structure is used in CUVIDPICPARAMS structure
/***********************************************************/
typedef struct _CUVIDVP9PICPARAMS
{
unsigned int width;
unsigned int height;
//Frame Indices
unsigned char LastRefIdx;
unsigned char GoldenRefIdx;
unsigned char AltRefIdx;
unsigned char colorSpace;
unsigned short profile : 3;
unsigned short frameContextIdx : 2;
unsigned short frameType : 1;
unsigned short showFrame : 1;
unsigned short errorResilient : 1;
unsigned short frameParallelDecoding : 1;
unsigned short subSamplingX : 1;
unsigned short subSamplingY : 1;
unsigned short intraOnly : 1;
unsigned short allow_high_precision_mv : 1;
unsigned short refreshEntropyProbs : 1;
unsigned short reserved2Bits : 2;
unsigned short reserved16Bits;
unsigned char refFrameSignBias[4];
unsigned char bitDepthMinus8Luma;
unsigned char bitDepthMinus8Chroma;
unsigned char loopFilterLevel;
unsigned char loopFilterSharpness;
unsigned char modeRefLfEnabled;
unsigned char log2_tile_columns;
unsigned char log2_tile_rows;
unsigned char segmentEnabled : 1;
unsigned char segmentMapUpdate : 1;
unsigned char segmentMapTemporalUpdate : 1;
unsigned char segmentFeatureMode : 1;
unsigned char reserved4Bits : 4;
unsigned char segmentFeatureEnable[8][4];
short segmentFeatureData[8][4];
unsigned char mb_segment_tree_probs[7];
unsigned char segment_pred_probs[3];
unsigned char reservedSegment16Bits[2];
int qpYAc;
int qpYDc;
int qpChDc;
int qpChAc;
unsigned int activeRefIdx[3];
unsigned int resetFrameContext;
unsigned int mcomp_filter_type;
unsigned int mbRefLfDelta[4];
unsigned int mbModeLfDelta[2];
unsigned int frameTagSize;
unsigned int offsetToDctParts;
unsigned int reserved128Bits[4];
} CUVIDVP9PICPARAMS;
/******************************************************************************************/
//! \struct CUVIDPICPARAMS
//! Picture parameters for decoding
//! This structure is used in cuvidDecodePicture API
//! IN for cuvidDecodePicture
/******************************************************************************************/
typedef struct _CUVIDPICPARAMS
{
int PicWidthInMbs; /**< IN: Coded frame size in macroblocks */
int FrameHeightInMbs; /**< IN: Coded frame height in macroblocks */
int CurrPicIdx; /**< IN: Output index of the current picture */
int field_pic_flag; /**< IN: 0=frame picture, 1=field picture */
int bottom_field_flag; /**< IN: 0=top field, 1=bottom field (ignored if field_pic_flag=0) */
int second_field; /**< IN: Second field of a complementary field pair */
// Bitstream data
unsigned int nBitstreamDataLen; /**< IN: Number of bytes in bitstream data buffer */
const unsigned char *pBitstreamData; /**< IN: Ptr to bitstream data for this picture (slice-layer) */
unsigned int nNumSlices; /**< IN: Number of slices in this picture */
const unsigned int *pSliceDataOffsets; /**< IN: nNumSlices entries, contains offset of each slice within
the bitstream data buffer */
int ref_pic_flag; /**< IN: This picture is a reference picture */
int intra_pic_flag; /**< IN: This picture is entirely intra coded */
unsigned int Reserved[30]; /**< Reserved for future use */
// IN: Codec-specific data
union {
CUVIDMPEG2PICPARAMS mpeg2; /**< Also used for MPEG-1 */
CUVIDH264PICPARAMS h264;
CUVIDVC1PICPARAMS vc1;
CUVIDMPEG4PICPARAMS mpeg4;
CUVIDJPEGPICPARAMS jpeg;
CUVIDHEVCPICPARAMS hevc;
CUVIDVP8PICPARAMS vp8;
CUVIDVP9PICPARAMS vp9;
unsigned int CodecReserved[1024];
} CodecSpecific;
} CUVIDPICPARAMS;
/******************************************************/
//! \struct CUVIDPROCPARAMS
//! Picture parameters for postprocessing
//! This structure is used in cuvidMapVideoFrame API
/******************************************************/
typedef struct _CUVIDPROCPARAMS
{
int progressive_frame; /**< IN: Input is progressive (deinterlace_mode will be ignored) */
int second_field; /**< IN: Output the second field (ignored if deinterlace mode is Weave) */
int top_field_first; /**< IN: Input frame is top field first (1st field is top, 2nd field is bottom) */
int unpaired_field; /**< IN: Input only contains one field (2nd field is invalid) */
// The fields below are used for raw YUV input
unsigned int reserved_flags; /**< Reserved for future use (set to zero) */
unsigned int reserved_zero; /**< Reserved (set to zero) */
unsigned long long raw_input_dptr; /**< IN: Input CUdeviceptr for raw YUV extensions */
unsigned int raw_input_pitch; /**< IN: pitch in bytes of raw YUV input (should be aligned appropriately) */
unsigned int raw_input_format; /**< IN: Input YUV format (cudaVideoCodec_enum) */
unsigned long long raw_output_dptr; /**< IN: Output CUdeviceptr for raw YUV extensions */
unsigned int raw_output_pitch; /**< IN: pitch in bytes of raw YUV output (should be aligned appropriately) */
unsigned int Reserved1; /**< Reserved for future use (set to zero) */
CUstream output_stream; /**< IN: stream object used by cuvidMapVideoFrame */
unsigned int Reserved[46]; /**< Reserved for future use (set to zero) */
void *Reserved2[2]; /**< Reserved for future use (set to zero) */
} CUVIDPROCPARAMS;
/***********************************************************************************************************/
//! VIDEO_DECODER
//!
//! In order to minimize decode latencies, there should be always at least 2 pictures in the decode
//! queue at any time, in order to make sure that all decode engines are always busy.
//!
//! Overall data flow:
//! - cuvidGetDecoderCaps(...)
//! - cuvidCreateDecoder(...)
//! - For each picture:
//! + cuvidDecodePicture(N)
//! + cuvidMapVideoFrame(N-4)
//! + do some processing in cuda
//! + cuvidUnmapVideoFrame(N-4)
//! + cuvidDecodePicture(N+1)
//! + cuvidMapVideoFrame(N-3)
//! + ...
//! - cuvidDestroyDecoder(...)
//!
//! NOTE:
//! - When the cuda context is created from a D3D device, the D3D device must also be created
//! with the D3DCREATE_MULTITHREADED flag.
//! - There is a limit to how many pictures can be mapped simultaneously (ulNumOutputSurfaces)
//! - cuvidDecodePicture may block the calling thread if there are too many pictures pending
//! in the decode queue
/***********************************************************************************************************/
/**********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidGetDecoderCaps(CUVIDDECODECAPS *pdc)
//! Queries decode capabilities of NVDEC-HW based on CodecType, ChromaFormat and BitDepthMinus8 parameters.
//! 1. Application fills IN parameters CodecType, ChromaFormat and BitDepthMinus8 of CUVIDDECODECAPS structure
//! 2. On calling cuvidGetDecoderCaps, driver fills OUT parameters if the IN parameters are supported
//! If IN parameters passed to the driver are not supported by NVDEC-HW, then all OUT params are set to 0.
//! E.g. on Geforce GTX 960:
//! App fills - eCodecType = cudaVideoCodec_H264; eChromaFormat = cudaVideoChromaFormat_420; nBitDepthMinus8 = 0;
//! Given IN parameters are supported, hence driver fills: bIsSupported = 1; nMinWidth = 48; nMinHeight = 16;
//! nMaxWidth = 4096; nMaxHeight = 4096; nMaxMBCount = 65536;
//! CodedWidth*CodedHeight/256 must be less than or equal to nMaxMBCount
/**********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidGetDecoderCaps(CUVIDDECODECAPS *pdc);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci)
//! Create the decoder object based on pdci. A handle to the created decoder is returned
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCreateDecoder(CUvideodecoder *phDecoder, CUVIDDECODECREATEINFO *pdci);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidDestroyDecoder(CUvideodecoder hDecoder)
//! Destroy the decoder object.
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidDestroyDecoder(CUvideodecoder hDecoder);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams)
//! Decode a single picture (field or frame)
//! Kicks off HW decoding
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidDecodePicture(CUvideodecoder hDecoder, CUVIDPICPARAMS *pPicParams);
#if !defined(__CUVID_DEVPTR64) || defined(__CUVID_INTERNAL)
/************************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx, unsigned int *pDevPtr,
//! unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
//! Post-process and map video frame corresponding to nPicIdx for use in cuda. Returns cuda device pointer and associated
//! pitch of the video frame
/************************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidMapVideoFrame(CUvideodecoder hDecoder, int nPicIdx,
unsigned int *pDevPtr, unsigned int *pPitch,
CUVIDPROCPARAMS *pVPP);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr)
//! Unmap a previously mapped video frame
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidUnmapVideoFrame(CUvideodecoder hDecoder, unsigned int DevPtr);
#endif
#if defined(_WIN64) || defined(__LP64__) || defined(__x86_64) || defined(AMD64) || defined(_M_AMD64)
/************************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr,
//! unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
//! Post-process and map video frame corresponding to nPicIdx for use in cuda. Returns cuda device pointer and associated
//! pitch of the video frame
/************************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidMapVideoFrame64(CUvideodecoder hDecoder, int nPicIdx, unsigned long long *pDevPtr,
unsigned int *pPitch, CUVIDPROCPARAMS *pVPP);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
//! Unmap a previously mapped video frame
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidUnmapVideoFrame64(CUvideodecoder hDecoder, unsigned long long DevPtr);
#if defined(__CUVID_DEVPTR64) && !defined(__CUVID_INTERNAL)
#define tcuvidMapVideoFrame tcuvidMapVideoFrame64
#define tcuvidUnmapVideoFrame tcuvidUnmapVideoFrame64
#endif
#endif
/********************************************************************************************************************/
//!
//! Context-locking: to facilitate multi-threaded implementations, the following 4 functions
//! provide a simple mutex-style host synchronization. If a non-NULL context is specified
//! in CUVIDDECODECREATEINFO, the codec library will acquire the mutex associated with the given
//! context before making any cuda calls.
//! A multi-threaded application could create a lock associated with a context handle so that
//! multiple threads can safely share the same cuda context:
//! - use cuCtxPopCurrent immediately after context creation in order to create a 'floating' context
//! that can be passed to cuvidCtxLockCreate.
//! - When using a floating context, all cuda calls should only be made within a cuvidCtxLock/cuvidCtxUnlock section.
//!
//! NOTE: This is a safer alternative to cuCtxPushCurrent and cuCtxPopCurrent, and is not related to video
//! decoder in any way (implemented as a critical section associated with cuCtx{Push|Pop}Current calls).
/********************************************************************************************************************/
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx)
//! This API is used to create CtxLock object
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCtxLockCreate(CUvideoctxlock *pLock, CUcontext ctx);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCtxLockDestroy(CUvideoctxlock lck)
//! This API is used to free CtxLock object
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCtxLockDestroy(CUvideoctxlock lck);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags)
//! This API is used to acquire ctxlock
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCtxLock(CUvideoctxlock lck, unsigned int reserved_flags);
/********************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags)
//! This API is used to release ctxlock
/********************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCtxUnlock(CUvideoctxlock lck, unsigned int reserved_flags);
/**********************************************************************************************/
#if defined(__cplusplus)
}
#endif /* __cplusplus */
#endif // __CUDA_VIDEO_H__

View File

@@ -1,33 +1,268 @@
/*
* This file is part of FFmpeg.
* This copyright notice applies to this header file only:
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
* Copyright (c) 2016
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
#ifndef COMPAT_CUDA_DYNLINK_LOADER_H
#define COMPAT_CUDA_DYNLINK_LOADER_H
#ifndef AV_COMPAT_CUDA_DYNLINK_LOADER_H
#define AV_COMPAT_CUDA_DYNLINK_LOADER_H
#include "libavutil/log.h"
#include "compat/cuda/dynlink_cuda.h"
#include "compat/cuda/dynlink_nvcuvid.h"
#include "compat/nvenc/nvEncodeAPI.h"
#include "compat/w32dlfcn.h"
#define FFNV_LOAD_FUNC(path) dlopen((path), RTLD_LAZY)
#define FFNV_SYM_FUNC(lib, sym) dlsym((lib), (sym))
#define FFNV_FREE_FUNC(lib) dlclose(lib)
#define FFNV_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_ERROR, msg, __VA_ARGS__)
#define FFNV_DEBUG_LOG_FUNC(logctx, msg, ...) av_log(logctx, AV_LOG_DEBUG, msg, __VA_ARGS__)
#include "libavutil/log.h"
#include "libavutil/error.h"
#include <ffnvcodec/dynlink_loader.h>
#if defined(_WIN32)
# define LIB_HANDLE HMODULE
#else
# define LIB_HANDLE void*
#endif
#if defined(_WIN32) || defined(__CYGWIN__)
# define CUDA_LIBNAME "nvcuda.dll"
# define NVCUVID_LIBNAME "nvcuvid.dll"
# if ARCH_X86_64
# define NVENC_LIBNAME "nvEncodeAPI64.dll"
# else
# define NVENC_LIBNAME "nvEncodeAPI.dll"
# endif
#else
# define CUDA_LIBNAME "libcuda.so.1"
# define NVCUVID_LIBNAME "libnvcuvid.so.1"
# define NVENC_LIBNAME "libnvidia-encode.so.1"
#endif
#define LOAD_LIBRARY(l, path) \
do { \
if (!((l) = dlopen(path, RTLD_LAZY))) { \
av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", path); \
ret = AVERROR_UNKNOWN; \
goto error; \
} \
av_log(NULL, AV_LOG_TRACE, "Loaded lib: %s\n", path); \
} while (0)
#define LOAD_SYMBOL(fun, tp, symbol) \
do { \
if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) { \
av_log(NULL, AV_LOG_ERROR, "Cannot load %s\n", symbol); \
ret = AVERROR_UNKNOWN; \
goto error; \
} \
av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
} while (0)
#define LOAD_SYMBOL_OPT(fun, tp, symbol) \
do { \
if (!((f->fun) = (tp*)dlsym(f->lib, symbol))) { \
av_log(NULL, AV_LOG_DEBUG, "Cannot load optional %s\n", symbol); \
} else { \
av_log(NULL, AV_LOG_TRACE, "Loaded sym: %s\n", symbol); \
} \
} while (0)
#define GENERIC_LOAD_FUNC_PREAMBLE(T, n, N) \
T *f; \
int ret; \
\
n##_free_functions(functions); \
\
f = *functions = av_mallocz(sizeof(*f)); \
if (!f) \
return AVERROR(ENOMEM); \
\
LOAD_LIBRARY(f->lib, N);
#define GENERIC_LOAD_FUNC_FINALE(n) \
return 0; \
error: \
n##_free_functions(functions); \
return ret;
#define GENERIC_FREE_FUNC() \
if (!functions) \
return; \
if (*functions && (*functions)->lib) \
dlclose((*functions)->lib); \
av_freep(functions);
#ifdef AV_COMPAT_DYNLINK_CUDA_H
typedef struct CudaFunctions {
tcuInit *cuInit;
tcuDeviceGetCount *cuDeviceGetCount;
tcuDeviceGet *cuDeviceGet;
tcuDeviceGetName *cuDeviceGetName;
tcuDeviceComputeCapability *cuDeviceComputeCapability;
tcuCtxCreate_v2 *cuCtxCreate;
tcuCtxPushCurrent_v2 *cuCtxPushCurrent;
tcuCtxPopCurrent_v2 *cuCtxPopCurrent;
tcuCtxDestroy_v2 *cuCtxDestroy;
tcuMemAlloc_v2 *cuMemAlloc;
tcuMemFree_v2 *cuMemFree;
tcuMemcpy2D_v2 *cuMemcpy2D;
tcuGetErrorName *cuGetErrorName;
tcuGetErrorString *cuGetErrorString;
LIB_HANDLE lib;
} CudaFunctions;
#else
typedef struct CudaFunctions CudaFunctions;
#endif
typedef struct CuvidFunctions {
tcuvidGetDecoderCaps *cuvidGetDecoderCaps;
tcuvidCreateDecoder *cuvidCreateDecoder;
tcuvidDestroyDecoder *cuvidDestroyDecoder;
tcuvidDecodePicture *cuvidDecodePicture;
tcuvidMapVideoFrame *cuvidMapVideoFrame;
tcuvidUnmapVideoFrame *cuvidUnmapVideoFrame;
tcuvidCtxLockCreate *cuvidCtxLockCreate;
tcuvidCtxLockDestroy *cuvidCtxLockDestroy;
tcuvidCtxLock *cuvidCtxLock;
tcuvidCtxUnlock *cuvidCtxUnlock;
tcuvidCreateVideoSource *cuvidCreateVideoSource;
tcuvidCreateVideoSourceW *cuvidCreateVideoSourceW;
tcuvidDestroyVideoSource *cuvidDestroyVideoSource;
tcuvidSetVideoSourceState *cuvidSetVideoSourceState;
tcuvidGetVideoSourceState *cuvidGetVideoSourceState;
tcuvidGetSourceVideoFormat *cuvidGetSourceVideoFormat;
tcuvidGetSourceAudioFormat *cuvidGetSourceAudioFormat;
tcuvidCreateVideoParser *cuvidCreateVideoParser;
tcuvidParseVideoData *cuvidParseVideoData;
tcuvidDestroyVideoParser *cuvidDestroyVideoParser;
LIB_HANDLE lib;
} CuvidFunctions;
typedef NVENCSTATUS NVENCAPI tNvEncodeAPICreateInstance(NV_ENCODE_API_FUNCTION_LIST *functionList);
typedef NVENCSTATUS NVENCAPI tNvEncodeAPIGetMaxSupportedVersion(uint32_t* version);
typedef struct NvencFunctions {
tNvEncodeAPICreateInstance *NvEncodeAPICreateInstance;
tNvEncodeAPIGetMaxSupportedVersion *NvEncodeAPIGetMaxSupportedVersion;
LIB_HANDLE lib;
} NvencFunctions;
#ifdef AV_COMPAT_DYNLINK_CUDA_H
static inline void cuda_free_functions(CudaFunctions **functions)
{
GENERIC_FREE_FUNC();
}
#endif
static inline void cuvid_free_functions(CuvidFunctions **functions)
{
GENERIC_FREE_FUNC();
}
static inline void nvenc_free_functions(NvencFunctions **functions)
{
GENERIC_FREE_FUNC();
}
#ifdef AV_COMPAT_DYNLINK_CUDA_H
static inline int cuda_load_functions(CudaFunctions **functions)
{
GENERIC_LOAD_FUNC_PREAMBLE(CudaFunctions, cuda, CUDA_LIBNAME);
LOAD_SYMBOL(cuInit, tcuInit, "cuInit");
LOAD_SYMBOL(cuDeviceGetCount, tcuDeviceGetCount, "cuDeviceGetCount");
LOAD_SYMBOL(cuDeviceGet, tcuDeviceGet, "cuDeviceGet");
LOAD_SYMBOL(cuDeviceGetName, tcuDeviceGetName, "cuDeviceGetName");
LOAD_SYMBOL(cuDeviceComputeCapability, tcuDeviceComputeCapability, "cuDeviceComputeCapability");
LOAD_SYMBOL(cuCtxCreate, tcuCtxCreate_v2, "cuCtxCreate_v2");
LOAD_SYMBOL(cuCtxPushCurrent, tcuCtxPushCurrent_v2, "cuCtxPushCurrent_v2");
LOAD_SYMBOL(cuCtxPopCurrent, tcuCtxPopCurrent_v2, "cuCtxPopCurrent_v2");
LOAD_SYMBOL(cuCtxDestroy, tcuCtxDestroy_v2, "cuCtxDestroy_v2");
LOAD_SYMBOL(cuMemAlloc, tcuMemAlloc_v2, "cuMemAlloc_v2");
LOAD_SYMBOL(cuMemFree, tcuMemFree_v2, "cuMemFree_v2");
LOAD_SYMBOL(cuMemcpy2D, tcuMemcpy2D_v2, "cuMemcpy2D_v2");
LOAD_SYMBOL(cuGetErrorName, tcuGetErrorName, "cuGetErrorName");
LOAD_SYMBOL(cuGetErrorString, tcuGetErrorString, "cuGetErrorString");
GENERIC_LOAD_FUNC_FINALE(cuda);
}
#endif
static inline int cuvid_load_functions(CuvidFunctions **functions)
{
GENERIC_LOAD_FUNC_PREAMBLE(CuvidFunctions, cuvid, NVCUVID_LIBNAME);
LOAD_SYMBOL_OPT(cuvidGetDecoderCaps, tcuvidGetDecoderCaps, "cuvidGetDecoderCaps");
LOAD_SYMBOL(cuvidCreateDecoder, tcuvidCreateDecoder, "cuvidCreateDecoder");
LOAD_SYMBOL(cuvidDestroyDecoder, tcuvidDestroyDecoder, "cuvidDestroyDecoder");
LOAD_SYMBOL(cuvidDecodePicture, tcuvidDecodePicture, "cuvidDecodePicture");
#ifdef __CUVID_DEVPTR64
LOAD_SYMBOL(cuvidMapVideoFrame, tcuvidMapVideoFrame, "cuvidMapVideoFrame64");
LOAD_SYMBOL(cuvidUnmapVideoFrame, tcuvidUnmapVideoFrame, "cuvidUnmapVideoFrame64");
#else
LOAD_SYMBOL(cuvidMapVideoFrame, tcuvidMapVideoFrame, "cuvidMapVideoFrame");
LOAD_SYMBOL(cuvidUnmapVideoFrame, tcuvidUnmapVideoFrame, "cuvidUnmapVideoFrame");
#endif
LOAD_SYMBOL(cuvidCtxLockCreate, tcuvidCtxLockCreate, "cuvidCtxLockCreate");
LOAD_SYMBOL(cuvidCtxLockDestroy, tcuvidCtxLockDestroy, "cuvidCtxLockDestroy");
LOAD_SYMBOL(cuvidCtxLock, tcuvidCtxLock, "cuvidCtxLock");
LOAD_SYMBOL(cuvidCtxUnlock, tcuvidCtxUnlock, "cuvidCtxUnlock");
LOAD_SYMBOL(cuvidCreateVideoSource, tcuvidCreateVideoSource, "cuvidCreateVideoSource");
LOAD_SYMBOL(cuvidCreateVideoSourceW, tcuvidCreateVideoSourceW, "cuvidCreateVideoSourceW");
LOAD_SYMBOL(cuvidDestroyVideoSource, tcuvidDestroyVideoSource, "cuvidDestroyVideoSource");
LOAD_SYMBOL(cuvidSetVideoSourceState, tcuvidSetVideoSourceState, "cuvidSetVideoSourceState");
LOAD_SYMBOL(cuvidGetVideoSourceState, tcuvidGetVideoSourceState, "cuvidGetVideoSourceState");
LOAD_SYMBOL(cuvidGetSourceVideoFormat, tcuvidGetSourceVideoFormat, "cuvidGetSourceVideoFormat");
LOAD_SYMBOL(cuvidGetSourceAudioFormat, tcuvidGetSourceAudioFormat, "cuvidGetSourceAudioFormat");
LOAD_SYMBOL(cuvidCreateVideoParser, tcuvidCreateVideoParser, "cuvidCreateVideoParser");
LOAD_SYMBOL(cuvidParseVideoData, tcuvidParseVideoData, "cuvidParseVideoData");
LOAD_SYMBOL(cuvidDestroyVideoParser, tcuvidDestroyVideoParser, "cuvidDestroyVideoParser");
GENERIC_LOAD_FUNC_FINALE(cuvid);
}
static inline int nvenc_load_functions(NvencFunctions **functions)
{
GENERIC_LOAD_FUNC_PREAMBLE(NvencFunctions, nvenc, NVENC_LIBNAME);
LOAD_SYMBOL(NvEncodeAPICreateInstance, tNvEncodeAPICreateInstance, "NvEncodeAPICreateInstance");
LOAD_SYMBOL(NvEncodeAPIGetMaxSupportedVersion, tNvEncodeAPIGetMaxSupportedVersion, "NvEncodeAPIGetMaxSupportedVersion");
GENERIC_LOAD_FUNC_FINALE(nvenc);
}
#undef GENERIC_LOAD_FUNC_PREAMBLE
#undef LOAD_LIBRARY
#undef LOAD_SYMBOL
#undef GENERIC_LOAD_FUNC_FINALE
#undef GENERIC_FREE_FUNC
#undef CUDA_LIBNAME
#undef NVCUVID_LIBNAME
#undef NVENC_LIBNAME
#undef LIB_HANDLE
#endif
#endif /* COMPAT_CUDA_DYNLINK_LOADER_H */

View File

@@ -0,0 +1,356 @@
/*
* This copyright notice applies to this header file only:
*
* Copyright (c) 2010-2017 NVIDIA Corporation
*
* Permission is hereby granted, free of charge, to any person
* obtaining a copy of this software and associated documentation
* files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use,
* copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the software, and to permit persons to whom the
* software is furnished to do so, subject to the following
* conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
* OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
* HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
/********************************************************************************************************************/
//! \file nvcuvid.h
//! NVDECODE API provides video decoding interface to NVIDIA GPU devices.
//! \date 2015-2017
//! This file contains the interface constants, structure definitions and function prototypes.
/********************************************************************************************************************/
#if !defined(__NVCUVID_H__)
#define __NVCUVID_H__
#include "compat/cuda/dynlink_cuviddec.h"
#if defined(__cplusplus)
extern "C" {
#endif /* __cplusplus */
/*********************************
** Initialization
*********************************/
CUresult CUDAAPI cuvidInit(unsigned int Flags);
/***********************************************/
//!
//! High-level helper APIs for video sources
//!
/***********************************************/
typedef void *CUvideosource;
typedef void *CUvideoparser;
typedef long long CUvideotimestamp;
/************************************************************************/
//! \enum cudaVideoState
//! Video source state enums
//! Used in cuvidSetVideoSourceState and cuvidGetVideoSourceState APIs
/************************************************************************/
typedef enum {
cudaVideoState_Error = -1, /**< Error state (invalid source) */
cudaVideoState_Stopped = 0, /**< Source is stopped (or reached end-of-stream) */
cudaVideoState_Started = 1 /**< Source is running and delivering data */
} cudaVideoState;
/************************************************************************/
//! \enum cudaAudioCodec
//! Audio compression enums
//! Used in CUAUDIOFORMAT structure
/************************************************************************/
typedef enum {
cudaAudioCodec_MPEG1=0, /**< MPEG-1 Audio */
cudaAudioCodec_MPEG2, /**< MPEG-2 Audio */
cudaAudioCodec_MP3, /**< MPEG-1 Layer III Audio */
cudaAudioCodec_AC3, /**< Dolby Digital (AC3) Audio */
cudaAudioCodec_LPCM, /**< PCM Audio */
cudaAudioCodec_AAC, /**< AAC Audio */
} cudaAudioCodec;
/************************************************************************************************/
//! \ingroup STRUCTS
//! \struct CUVIDEOFORMAT
//! Video format
//! Used in cuvidGetSourceVideoFormat API
/************************************************************************************************/
typedef struct
{
cudaVideoCodec codec; /**< OUT: Compression format */
/**
* OUT: frame rate = numerator / denominator (for example: 30000/1001)
*/
struct {
/**< OUT: frame rate numerator (0 = unspecified or variable frame rate) */
unsigned int numerator;
/**< OUT: frame rate denominator (0 = unspecified or variable frame rate) */
unsigned int denominator;
} frame_rate;
unsigned char progressive_sequence; /**< OUT: 0=interlaced, 1=progressive */
unsigned char bit_depth_luma_minus8; /**< OUT: high bit depth luma. E.g, 2 for 10-bitdepth, 4 for 12-bitdepth */
unsigned char bit_depth_chroma_minus8; /**< OUT: high bit depth chroma. E.g, 2 for 10-bitdepth, 4 for 12-bitdepth */
unsigned char reserved1; /**< Reserved for future use */
unsigned int coded_width; /**< OUT: coded frame width in pixels */
unsigned int coded_height; /**< OUT: coded frame height in pixels */
/**
* area of the frame that should be displayed
* typical example:
* coded_width = 1920, coded_height = 1088
* display_area = { 0,0,1920,1080 }
*/
struct {
int left; /**< OUT: left position of display rect */
int top; /**< OUT: top position of display rect */
int right; /**< OUT: right position of display rect */
int bottom; /**< OUT: bottom position of display rect */
} display_area;
cudaVideoChromaFormat chroma_format; /**< OUT: Chroma format */
unsigned int bitrate; /**< OUT: video bitrate (bps, 0=unknown) */
/**
* OUT: Display Aspect Ratio = x:y (4:3, 16:9, etc)
*/
struct {
int x;
int y;
} display_aspect_ratio;
/**
* Video Signal Description
* Refer section E.2.1 (VUI parameters semantics) of H264 spec file
*/
struct {
unsigned char video_format : 3; /**< OUT: 0-Component, 1-PAL, 2-NTSC, 3-SECAM, 4-MAC, 5-Unspecified */
unsigned char video_full_range_flag : 1; /**< OUT: indicates the black level and luma and chroma range */
unsigned char reserved_zero_bits : 4; /**< Reserved bits */
unsigned char color_primaries; /**< OUT: chromaticity coordinates of source primaries */
unsigned char transfer_characteristics; /**< OUT: opto-electronic transfer characteristic of the source picture */
unsigned char matrix_coefficients; /**< OUT: used in deriving luma and chroma signals from RGB primaries */
} video_signal_description;
unsigned int seqhdr_data_length; /**< OUT: Additional bytes following (CUVIDEOFORMATEX) */
} CUVIDEOFORMAT;
/****************************************************************/
//! \ingroup STRUCTS
//! \struct CUVIDEOFORMATEX
//! Video format including raw sequence header information
//! Used in cuvidGetSourceVideoFormat API
/****************************************************************/
typedef struct
{
CUVIDEOFORMAT format; /**< OUT: CUVIDEOFORMAT structure */
unsigned char raw_seqhdr_data[1024]; /**< OUT: Sequence header data */
} CUVIDEOFORMATEX;
/****************************************************************/
//! \ingroup STRUCTS
//! \struct CUAUDIOFORMAT
//! Audio formats
//! Used in cuvidGetSourceAudioFormat API
/****************************************************************/
typedef struct
{
cudaAudioCodec codec; /**< OUT: Compression format */
unsigned int channels; /**< OUT: number of audio channels */
unsigned int samplespersec; /**< OUT: sampling frequency */
unsigned int bitrate; /**< OUT: For uncompressed, can also be used to determine bits per sample */
unsigned int reserved1; /**< Reserved for future use */
unsigned int reserved2; /**< Reserved for future use */
} CUAUDIOFORMAT;
/***************************************************************/
//! \enum CUvideopacketflags
//! Data packet flags
//! Used in CUVIDSOURCEDATAPACKET structure
/***************************************************************/
typedef enum {
CUVID_PKT_ENDOFSTREAM = 0x01, /**< Set when this is the last packet for this stream */
CUVID_PKT_TIMESTAMP = 0x02, /**< Timestamp is valid */
CUVID_PKT_DISCONTINUITY = 0x04, /**< Set when a discontinuity has to be signalled */
CUVID_PKT_ENDOFPICTURE = 0x08, /**< Set when the packet contains exactly one frame */
} CUvideopacketflags;
/*****************************************************************************/
//! \ingroup STRUCTS
//! \struct CUVIDSOURCEDATAPACKET
//! Data Packet
//! Used in cuvidParseVideoData API
//! IN for cuvidParseVideoData
/*****************************************************************************/
typedef struct _CUVIDSOURCEDATAPACKET
{
tcu_ulong flags; /**< IN: Combination of CUVID_PKT_XXX flags */
tcu_ulong payload_size; /**< IN: number of bytes in the payload (may be zero if EOS flag is set) */
const unsigned char *payload; /**< IN: Pointer to packet payload data (may be NULL if EOS flag is set) */
CUvideotimestamp timestamp; /**< IN: Presentation time stamp (10MHz clock), only valid if
CUVID_PKT_TIMESTAMP flag is set */
} CUVIDSOURCEDATAPACKET;
// Callback for packet delivery
typedef int (CUDAAPI *PFNVIDSOURCECALLBACK)(void *, CUVIDSOURCEDATAPACKET *);
/**************************************************************************************************************************/
//! \ingroup STRUCTS
//! \struct CUVIDSOURCEPARAMS
//! Describes parameters needed in cuvidCreateVideoSource API
//! NVDECODE API is intended for HW accelerated video decoding so CUvideosource doesn't have audio demuxer for all supported
//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
/**************************************************************************************************************************/
typedef struct _CUVIDSOURCEPARAMS
{
unsigned int ulClockRate; /**< IN: Time stamp units in Hz (0=default=10000000Hz) */
unsigned int uReserved1[7]; /**< Reserved for future use - set to zero */
void *pUserData; /**< IN: User private data passed in to the data handlers */
PFNVIDSOURCECALLBACK pfnVideoDataHandler; /**< IN: Called to deliver video packets */
PFNVIDSOURCECALLBACK pfnAudioDataHandler; /**< IN: Called to deliver audio packets. */
void *pvReserved2[8]; /**< Reserved for future use - set to NULL */
} CUVIDSOURCEPARAMS;
/**********************************************/
//! \ingroup ENUMS
//! \enum CUvideosourceformat_flags
//! CUvideosourceformat_flags
//! Used in cuvidGetSourceVideoFormat API
/**********************************************/
typedef enum {
CUVID_FMT_EXTFORMATINFO = 0x100 /**< Return extended format structure (CUVIDEOFORMATEX) */
} CUvideosourceformat_flags;
#if !defined(__APPLE__)
/**************************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams)
//! Create CUvideosource object. CUvideosource spawns demultiplexer thread that provides two callbacks:
//! pfnVideoDataHandler() and pfnAudioDataHandler()
//! NVDECODE API is intended for HW accelerated video decoding so CUvideosource doesn't have audio demuxer for all supported
//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
/**************************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCreateVideoSource(CUvideosource *pObj, const char *pszFileName, CUVIDSOURCEPARAMS *pParams);
/****************************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams)
//! Create video source object and initialize
/****************************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCreateVideoSourceW(CUvideosource *pObj, const wchar_t *pwszFileName, CUVIDSOURCEPARAMS *pParams);
/*********************************************************************/
//! \fn CUresult CUDAAPI cuvidDestroyVideoSource(CUvideosource obj)
//! Destroy video source
/*********************************************************************/
typedef CUresult CUDAAPI tcuvidDestroyVideoSource(CUvideosource obj);
/******************************************************************************************/
//! \fn CUresult CUDAAPI cuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state)
//! Set video source state
/******************************************************************************************/
typedef CUresult CUDAAPI tcuvidSetVideoSourceState(CUvideosource obj, cudaVideoState state);
/******************************************************************************************/
//! \fn cudaVideoState CUDAAPI cuvidGetVideoSourceState(CUvideosource obj)
//! Get video source state
/******************************************************************************************/
typedef cudaVideoState CUDAAPI tcuvidGetVideoSourceState(CUvideosource obj);
/****************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags)
//! Gets details of video stream in pvidfmt
/****************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidGetSourceVideoFormat(CUvideosource obj, CUVIDEOFORMAT *pvidfmt, unsigned int flags);
/****************************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags)
//! Get audio source format
//! NVDECODE API is intended for HW accelarated video decoding so CUvideosource doesn't have audio demuxer for all suppported
//! containers. It's recommended to clients to use their own or third party demuxer if audio support is needed.
/****************************************************************************************************************/
typedef CUresult CUDAAPI tcuvidGetSourceAudioFormat(CUvideosource obj, CUAUDIOFORMAT *paudfmt, unsigned int flags);
#endif
/**********************************************************************************/
//! \ingroup STRUCTS
//! \struct CUVIDPARSERDISPINFO
//! Used in cuvidParseVideoData API with PFNVIDDISPLAYCALLBACK pfnDisplayPicture
/**********************************************************************************/
typedef struct _CUVIDPARSERDISPINFO
{
int picture_index; /**< OUT: Index of the current picture */
int progressive_frame; /**< OUT: 1 if progressive frame; 0 otherwise */
int top_field_first; /**< OUT: 1 if top field is displayed first; 0 otherwise */
int repeat_first_field; /**< OUT: Number of additional fields (1=ivtc, 2=frame doubling, 4=frame tripling,
-1=unpaired field) */
CUvideotimestamp timestamp; /**< OUT: Presentation time stamp */
} CUVIDPARSERDISPINFO;
/***********************************************************************************************************************/
//! Parser callbacks
//! The parser will call these synchronously from within cuvidParseVideoData(), whenever a picture is ready to
//! be decoded and/or displayed. First argument in functions is "void *pUserData" member of structure CUVIDSOURCEPARAMS
/***********************************************************************************************************************/
typedef int (CUDAAPI *PFNVIDSEQUENCECALLBACK)(void *, CUVIDEOFORMAT *);
typedef int (CUDAAPI *PFNVIDDECODECALLBACK)(void *, CUVIDPICPARAMS *);
typedef int (CUDAAPI *PFNVIDDISPLAYCALLBACK)(void *, CUVIDPARSERDISPINFO *);
/**************************************/
//! \ingroup STRUCTS
//! \struct CUVIDPARSERPARAMS
//! Used in cuvidCreateVideoParser API
/**************************************/
typedef struct _CUVIDPARSERPARAMS
{
cudaVideoCodec CodecType; /**< IN: cudaVideoCodec_XXX */
unsigned int ulMaxNumDecodeSurfaces; /**< IN: Max # of decode surfaces (parser will cycle through these) */
unsigned int ulClockRate; /**< IN: Timestamp units in Hz (0=default=10000000Hz) */
unsigned int ulErrorThreshold; /**< IN: % Error threshold (0-100) for calling pfnDecodePicture (100=always
IN: call pfnDecodePicture even if picture bitstream is fully corrupted) */
unsigned int ulMaxDisplayDelay; /**< IN: Max display queue delay (improves pipelining of decode with display)
0=no delay (recommended values: 2..4) */
unsigned int uReserved1[5]; /**< IN: Reserved for future use - set to 0 */
void *pUserData; /**< IN: User data for callbacks */
PFNVIDSEQUENCECALLBACK pfnSequenceCallback; /**< IN: Called before decoding frames and/or whenever there is a fmt change */
PFNVIDDECODECALLBACK pfnDecodePicture; /**< IN: Called when a picture is ready to be decoded (decode order) */
PFNVIDDISPLAYCALLBACK pfnDisplayPicture; /**< IN: Called whenever a picture is ready to be displayed (display order) */
void *pvReserved2[7]; /**< Reserved for future use - set to NULL */
CUVIDEOFORMATEX *pExtVideoInfo; /**< IN: [Optional] sequence header data from system layer */
} CUVIDPARSERPARAMS;
/************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams)
//! Create video parser object and initialize
/************************************************************************************************/
typedef CUresult CUDAAPI tcuvidCreateVideoParser(CUvideoparser *pObj, CUVIDPARSERPARAMS *pParams);
/************************************************************************************************/
//! \fn CUresult CUDAAPI cuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket)
//! Parse the video data from source data packet in pPacket
//! Extracts parameter sets like SPS, PPS, bitstream etc. from pPacket and
//! calls back pfnDecodePicture with CUVIDPICPARAMS data for kicking of HW decoding
/************************************************************************************************/
typedef CUresult CUDAAPI tcuvidParseVideoData(CUvideoparser obj, CUVIDSOURCEDATAPACKET *pPacket);
/*******************************************************************/
//! \fn CUresult CUDAAPI cuvidDestroyVideoParser(CUvideoparser obj)
/*******************************************************************/
typedef CUresult CUDAAPI tcuvidDestroyVideoParser(CUvideoparser obj);
/**********************************************************************************************/
#if defined(__cplusplus)
}
#endif /* __cplusplus */
#endif // __NVCUVID_H__

View File

@@ -27,8 +27,10 @@ IN="$2"
NAME="$(basename "$IN" | sed 's/\..*//')"
printf "const char %s_ptx[] = \\" "$NAME" > "$OUT"
echo >> "$OUT"
sed -e "$(printf 's/\r//g')" -e 's/["\\]/\\&/g' -e "$(printf 's/^/\t"/')" -e 's/$/\\n"/' < "$IN" >> "$OUT"
echo ";" >> "$OUT"
while read LINE
do
printf "\n\t\"%s\\\n\"" "$(printf "%s" "$LINE" | sed -e 's/\r//g' -e 's/["\\]/\\&/g')" >> "$OUT"
done < "$IN"
printf ";\n" >> "$OUT"
exit 0

View File

@@ -1,47 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
#define FUN(name, type, op) \
type name(type x, type y) \
{ \
if (fpclassify(x) == FP_NAN) return y; \
if (fpclassify(y) == FP_NAN) return x; \
return x op y ? x : y; \
}
FUN(fmin, double, <)
FUN(fmax, double, >)
FUN(fminf, float, <)
FUN(fmaxf, float, >)
long double fmodl(long double x, long double y)
{
return fmod(x, y);
}
long double scalbnl(long double x, int exp)
{
return scalbn(x, exp);
}
long double copysignl(long double x, long double y)
{
return copysign(x, y);
}

3324
compat/nvenc/nvEncodeAPI.h Normal file

File diff suppressed because it is too large Load Diff

View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011-2017 KO Myung-Hun <komh@chollian.net>
* Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
*
* This file is part of FFmpeg.
*
@@ -27,19 +27,15 @@
#define COMPAT_OS2THREADS_H
#define INCL_DOS
#define INCL_DOSERRORS
#include <os2.h>
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#include <time.h>
#include <sys/builtin.h>
#include <sys/fmutex.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/time.h"
typedef struct {
TID tid;
@@ -50,11 +46,9 @@ typedef struct {
typedef void pthread_attr_t;
typedef _fmutex pthread_mutex_t;
typedef HMTX pthread_mutex_t;
typedef void pthread_mutexattr_t;
#define PTHREAD_MUTEX_INITIALIZER _FMUTEX_INITIALIZER
typedef struct {
HEV event_sem;
HEV ack_sem;
@@ -104,28 +98,28 @@ static av_always_inline int pthread_join(pthread_t thread, void **value_ptr)
static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex,
const pthread_mutexattr_t *attr)
{
_fmutex_create(mutex, 0);
DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE);
return 0;
}
static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex)
{
_fmutex_close(mutex);
DosCloseMutexSem(*(PHMTX)mutex);
return 0;
}
static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex)
{
_fmutex_request(mutex, 0);
DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT);
return 0;
}
static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex)
{
_fmutex_release(mutex);
DosReleaseMutexSem(*(PHMTX)mutex);
return 0;
}
@@ -167,28 +161,6 @@ static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond)
return 0;
}
static av_always_inline int pthread_cond_timedwait(pthread_cond_t *cond,
pthread_mutex_t *mutex,
const struct timespec *abstime)
{
int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
ULONG t = av_clip64(abs_milli - av_gettime() / 1000, 0, ULONG_MAX);
__atomic_increment(&cond->wait_count);
pthread_mutex_unlock(mutex);
APIRET ret = DosWaitEventSem(cond->event_sem, t);
__atomic_decrement(&cond->wait_count);
DosPostEventSem(cond->ack_sem);
pthread_mutex_lock(mutex);
return (ret == ERROR_TIMEOUT) ? ETIMEDOUT : 0;
}
static av_always_inline int pthread_cond_wait(pthread_cond_t *cond,
pthread_mutex_t *mutex)
{

30
compat/tms470/math.h Normal file
View File

@@ -0,0 +1,30 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef COMPAT_TMS470_MATH_H
#define COMPAT_TMS470_MATH_H
#include_next <math.h>
#undef INFINITY
#undef NAN
#define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
#define NAN (*(const float*)((const unsigned []){ 0x7fc00000 }))
#endif /* COMPAT_TMS470_MATH_H */

View File

@@ -21,7 +21,6 @@
#ifdef _WIN32
#include <windows.h>
#include "config.h"
#if (_WIN32_WINNT < 0x0602) || HAVE_WINRT
#include "libavutil/wchar_filename.h"
#endif

View File

@@ -38,13 +38,16 @@
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <process.h>
#include <time.h>
#if _WIN32_WINNT < 0x0600 && defined(__MINGW32__)
#undef MemoryBarrier
#define MemoryBarrier __sync_synchronize
#endif
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "libavutil/internal.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
typedef struct pthread_t {
void *handle;
@@ -53,18 +56,24 @@ typedef struct pthread_t {
void *ret;
} pthread_t;
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
/* the conditional variable api for windows 6.0+ uses critical sections and
* not mutexes */
typedef CRITICAL_SECTION pthread_mutex_t;
/* This is the CONDITION_VARIABLE typedef for using Windows' native
* conditional variables on kernels 6.0+. */
#if HAVE_CONDITION_VARIABLE_PTR
typedef CONDITION_VARIABLE pthread_cond_t;
#else
typedef struct pthread_cond_t {
void *Ptr;
} pthread_cond_t;
#endif
#define PTHREAD_MUTEX_INITIALIZER SRWLOCK_INIT
#define PTHREAD_COND_INITIALIZER CONDITION_VARIABLE_INIT
#if _WIN32_WINNT >= 0x0600
#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
#define PTHREAD_CANCEL_ENABLE 1
#define PTHREAD_CANCEL_DISABLE 0
#endif
static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
{
@@ -105,25 +114,26 @@ static av_unused int pthread_join(pthread_t thread, void **value_ptr)
static inline int pthread_mutex_init(pthread_mutex_t *m, void* attr)
{
InitializeSRWLock(m);
InitializeCriticalSection(m);
return 0;
}
static inline int pthread_mutex_destroy(pthread_mutex_t *m)
{
/* Unlocked SWR locks use no resources */
DeleteCriticalSection(m);
return 0;
}
static inline int pthread_mutex_lock(pthread_mutex_t *m)
{
AcquireSRWLockExclusive(m);
EnterCriticalSection(m);
return 0;
}
static inline int pthread_mutex_unlock(pthread_mutex_t *m)
{
ReleaseSRWLockExclusive(m);
LeaveCriticalSection(m);
return 0;
}
#if _WIN32_WINNT >= 0x0600
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
@@ -157,23 +167,7 @@ static inline int pthread_cond_broadcast(pthread_cond_t *cond)
static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
SleepConditionVariableSRW(cond, mutex, INFINITE, 0);
return 0;
}
static inline int pthread_cond_timedwait(pthread_cond_t *cond, pthread_mutex_t *mutex,
const struct timespec *abstime)
{
int64_t abs_milli = abstime->tv_sec * 1000LL + abstime->tv_nsec / 1000000;
DWORD t = av_clip64(abs_milli - av_gettime() / 1000, 0, UINT32_MAX);
if (!SleepConditionVariableSRW(cond, mutex, t, 0)) {
DWORD err = GetLastError();
if (err == ERROR_TIMEOUT)
return ETIMEDOUT;
else
return EINVAL;
}
SleepConditionVariableCS(cond, mutex, INFINITE);
return 0;
}
@@ -183,9 +177,242 @@ static inline int pthread_cond_signal(pthread_cond_t *cond)
return 0;
}
static inline int pthread_setcancelstate(int state, int *oldstate)
#else // _WIN32_WINNT < 0x0600
/* atomic init state of dynamically loaded functions */
static LONG w32thread_init_state = 0;
static av_unused void w32thread_init(void);
/* for pre-Windows 6.0 platforms, define INIT_ONCE struct,
* compatible to the one used in the native API */
typedef union pthread_once_t {
void * Ptr; ///< For the Windows 6.0+ native functions
LONG state; ///< For the pre-Windows 6.0 compat code
} pthread_once_t;
#define PTHREAD_ONCE_INIT {0}
/* function pointers to init once API on windows 6.0+ kernels */
static BOOL (WINAPI *initonce_begin)(pthread_once_t *lpInitOnce, DWORD dwFlags, BOOL *fPending, void **lpContext);
static BOOL (WINAPI *initonce_complete)(pthread_once_t *lpInitOnce, DWORD dwFlags, void *lpContext);
/* pre-Windows 6.0 compat using a spin-lock */
static inline void w32thread_once_fallback(LONG volatile *state, void (*init_routine)(void))
{
switch (InterlockedCompareExchange(state, 1, 0)) {
/* Initial run */
case 0:
init_routine();
InterlockedExchange(state, 2);
break;
/* Another thread is running init */
case 1:
while (1) {
MemoryBarrier();
if (*state == 2)
break;
Sleep(0);
}
break;
/* Initialization complete */
case 2:
break;
}
}
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
/* Use native functions on Windows 6.0+ */
if (initonce_begin && initonce_complete) {
BOOL pending = FALSE;
initonce_begin(once_control, 0, &pending, NULL);
if (pending)
init_routine();
initonce_complete(once_control, 0, NULL);
return 0;
}
w32thread_once_fallback(&once_control->state, init_routine);
return 0;
}
/* for pre-Windows 6.0 platforms we need to define and use our own condition
* variable and api */
typedef struct win32_cond_t {
pthread_mutex_t mtx_broadcast;
pthread_mutex_t mtx_waiter_count;
volatile int waiter_count;
HANDLE semaphore;
HANDLE waiters_done;
volatile int is_broadcast;
} win32_cond_t;
/* function pointers to conditional variable API on windows 6.0+ kernels */
static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
static void (WINAPI *cond_init)(pthread_cond_t *cond);
static void (WINAPI *cond_signal)(pthread_cond_t *cond);
static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
DWORD milliseconds);
static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
{
win32_cond_t *win32_cond = NULL;
w32thread_once_fallback(&w32thread_init_state, w32thread_init);
if (cond_init) {
cond_init(cond);
return 0;
}
/* non native condition variables */
win32_cond = (win32_cond_t*)av_mallocz(sizeof(win32_cond_t));
if (!win32_cond)
return ENOMEM;
cond->Ptr = win32_cond;
win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
if (!win32_cond->semaphore)
return ENOMEM;
win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
if (!win32_cond->waiters_done)
return ENOMEM;
pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
return 0;
}
static av_unused int pthread_cond_destroy(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
/* native condition variables do not destroy */
if (cond_init)
return 0;
/* non native condition variables */
CloseHandle(win32_cond->semaphore);
CloseHandle(win32_cond->waiters_done);
pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
pthread_mutex_destroy(&win32_cond->mtx_broadcast);
av_freep(&win32_cond);
cond->Ptr = NULL;
return 0;
}
static av_unused int pthread_cond_broadcast(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int have_waiter;
if (cond_broadcast) {
cond_broadcast(cond);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = 0;
if (win32_cond->waiter_count) {
win32_cond->is_broadcast = 1;
have_waiter = 1;
}
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, win32_cond->waiter_count, NULL);
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
win32_cond->is_broadcast = 0;
} else
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int last_waiter;
if (cond_wait) {
cond_wait(cond, mutex, INFINITE);
return 0;
}
/* non native condition variables */
pthread_mutex_lock(&win32_cond->mtx_broadcast);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count++;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
// unlock the external mutex
pthread_mutex_unlock(mutex);
WaitForSingleObject(win32_cond->semaphore, INFINITE);
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
win32_cond->waiter_count--;
last_waiter = !win32_cond->waiter_count || !win32_cond->is_broadcast;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (last_waiter)
SetEvent(win32_cond->waiters_done);
// lock the external mutex
return pthread_mutex_lock(mutex);
}
static av_unused int pthread_cond_signal(pthread_cond_t *cond)
{
win32_cond_t *win32_cond = (win32_cond_t*)cond->Ptr;
int have_waiter;
if (cond_signal) {
cond_signal(cond);
return 0;
}
pthread_mutex_lock(&win32_cond->mtx_broadcast);
/* non-native condition variables */
pthread_mutex_lock(&win32_cond->mtx_waiter_count);
have_waiter = win32_cond->waiter_count;
pthread_mutex_unlock(&win32_cond->mtx_waiter_count);
if (have_waiter) {
ReleaseSemaphore(win32_cond->semaphore, 1, NULL);
WaitForSingleObject(win32_cond->waiters_done, INFINITE);
ResetEvent(win32_cond->waiters_done);
}
pthread_mutex_unlock(&win32_cond->mtx_broadcast);
return 0;
}
#endif
static av_unused void w32thread_init(void)
{
#if _WIN32_WINNT < 0x0600
HMODULE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
/* if one is available, then they should all be available */
cond_init = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "InitializeConditionVariable");
cond_broadcast = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "WakeAllConditionVariable");
cond_signal = (void (WINAPI*)(pthread_cond_t *))
GetProcAddress(kernel_dll, "WakeConditionVariable");
cond_wait = (BOOL (WINAPI*)(pthread_cond_t *, pthread_mutex_t *, DWORD))
GetProcAddress(kernel_dll, "SleepConditionVariableCS");
initonce_begin = (BOOL (WINAPI*)(pthread_once_t *, DWORD, BOOL *, void **))
GetProcAddress(kernel_dll, "InitOnceBeginInitialize");
initonce_complete = (BOOL (WINAPI*)(pthread_once_t *, DWORD, void *))
GetProcAddress(kernel_dll, "InitOnceComplete");
#endif
}
#endif /* COMPAT_W32PTHREADS_H */

View File

@@ -45,11 +45,7 @@ libname=$(mktemp -u "library").lib
trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
lib.exe -out:${libname} $@ >/dev/null
fi
lib -out:${libname} $@ >/dev/null
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
exit 1
@@ -58,7 +54,23 @@ fi
IFS='
'
prefix="$EXTERN_PREFIX"
# Determine if we're building for x86 or x86_64 and
# set the symbol prefix accordingly.
prefix=""
arch=$(dumpbin -headers ${libname} |
tr '\t' ' ' |
grep '^ \+.\+machine \+(.\+)' |
head -1 |
sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
if [ "${arch}" = "x86" ]; then
prefix="_"
else
if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
echo "Unknown machine type." >&2
exit 1
fi
fi
started=0
regex="none"
@@ -100,19 +112,7 @@ for line in $(cat ${vscript} | tr '\t' ' '); do
'
done
if [ -n "$NM" ]; then
# Use eval, since NM="nm -g"
dump=$(eval "$NM --defined-only -g ${libname}" |
grep -v : |
grep -v ^$ |
cut -d' ' -f3 |
sed -e "s/^${prefix}//")
else
dump=$(dumpbin.exe -linkermember:1 ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3)
fi
dump=$(dumpbin -linkermember:1 ${libname})
rm ${libname}
@@ -121,6 +121,9 @@ list=""
for exp in ${regex}; do
list="${list}"'
'$(echo "${dump}" |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
cut -d' ' -f3 |
grep "^${exp}" |
sed -e 's/^/ /')
done

View File

@@ -4,6 +4,6 @@ LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
if [ -x "$LINK_EXE_PATH" ]; then
"$LINK_EXE_PATH" $@
else
link.exe $@
link $@
fi
exit $?

3168
configure vendored

File diff suppressed because it is too large Load Diff

View File

@@ -2,474 +2,19 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
since the last major version increase or the API was added.
The last version increases were:
libavcodec: 2017-10-21
libavdevice: 2017-10-21
libavfilter: 2017-10-21
libavformat: 2017-10-21
libavresample: 2017-10-21
libpostproc: 2017-10-21
libswresample: 2017-10-21
libswscale: 2017-10-21
libavutil: 2017-10-21
libavcodec: 2015-08-28
libavdevice: 2015-08-28
libavfilter: 2015-08-28
libavformat: 2015-08-28
libavresample: 2015-08-28
libpostproc: 2015-08-28
libswresample: 2015-08-28
libswscale: 2015-08-28
libavutil: 2015-08-28
API changes, most recent first:
-------- 8< --------- FFmpeg 4.4 was cut here -------- 8< ---------
2021-03-19 - e8c0bca6bd - lavu 56.69.100 - adler32.h
Added a typedef for the type of the Adler-32 checksums
used by av_adler32_update(). It will be changed to uint32_t
at the next major bump.
The type of the parameter for the length of the input buffer
will also be changed to size_t at the next major bump.
2021-03-19 - e318438f2f - lavf 58.75.100 - avformat.h
AVChapter.id will be changed from int to int64_t
on the next major version bump.
2021-03-17 - f7db77bd87 - lavc 58.133.100 - codec.h
Deprecated av_init_packet(). Once removed, sizeof(AVPacket) will
no longer be a part of the public ABI.
Deprecated AVPacketList.
2021-03-16 - 7d09579190 - lavc 58.132.100 - codec.h
Add AV_CODEC_CAP_OTHER_THREADS as a new name for
AV_CODEC_CAP_AUTO_THREADS. AV_CODEC_CAP_AUTO_THREADS
is now deprecated.
2021-03-12 - 6e7e3a3820 - lavc 58.131.100 - avcodec.h codec.h
Add a get_encode_buffer callback to AVCodecContext, similar to
get_buffer2 but for encoders.
Add avcodec_default_get_encode_buffer().
Add AV_GET_ENCODE_BUFFER_FLAG_REF.
Encoders may now be flagged as AV_CODEC_CAP_DR1 capable.
2021-03-10 - 42e68fe015 - lavf 58.72.100 - avformat.h
Change AVBufferRef related AVStream function and struct size
parameter and fields type to size_t at next major bump.
2021-03-10 - d79e0fe65c - lavc 58.130.100 - packet.h
Change AVBufferRef related AVPacket function and struct size
parameter and fields type to size_t at next major bump.
2021-03-10 - 14040a1d91 - lavu 56.68.100 - buffer.h frame.h
Change AVBufferRef and relevant AVFrame function and struct size
parameter and fields type to size_t at next major bump.
2021-03-04 - a0eec776b6 - lavc 58.128.101 - avcodec.h
Enable err_recognition to be set for encoders.
2021-03-03 - 2ff40b98ec - lavf 58.70.100 - avformat.h
Deprecate AVFMT_FLAG_PRIV_OPT. It will do nothing
as soon as av_demuxer_open() is removed.
2021-02-27 - dd9227e48f - lavc 58.126.100 - avcodec.h
Deprecated avcodec_get_frame_class().
2021-02-21 - 5ca40d6d94 - lavu 56.66.100 - tx.h
Add enum AVTXFlags and AVTXFlags.AV_TX_INPLACE
2021-02-14 - 4f49ca7bbc - lavd 58.12.100 - avdevice.h
Deprecated avdevice_capabilities_create() and
avdevice_capabilities_free().
2021-02-10 - 1bda9bb68a - lavu 56.65.100 - common.h
Add FFABS64U()
2021-01-26 - 5dd9567080 - lavu 56.64.100 - common.h
Add FFABSU()
2021-01-25 - 56709ca8aa - lavc 58.119.100 - avcodec.h
Deprecate AVCodecContext.debug_mv, FF_DEBUG_VIS_MV_P_FOR, FF_DEBUG_VIS_MV_B_FOR,
FF_DEBUG_VIS_MV_B_BACK
2021-01-11 - ebdd33086a - lavc 58.116.100 - avcodec.h
Add FF_PROFILE_VVC_MAIN_10 and FF_PROFILE_VVC_MAIN_10_444.
2020-01-01 - baecaa16c1 - lavu 56.63.100 - video_enc_params.h
Add AV_VIDEO_ENC_PARAMS_MPEG2
2020-12-03 - eca12f4d5a - lavu 56.62.100 - timecode.h
Add av_timecode_init_from_components.
2020-11-27 - a83098ab03 - lavc 58.114.100 - avcodec.h
Deprecate AVCodecContext.thread_safe_callbacks. Starting with
LIBAVCODEC_VERSION_MAJOR=60, user callbacks must always be
thread-safe when frame threading is used.
2020-11-25 - d243dd540a - lavc 58.113.100 - avcodec.h
Adds a new flag AV_CODEC_EXPORT_DATA_FILM_GRAIN for export_side_data.
2020-11-25 - 4f9ee87253 - lavu 56.61.100 - film_grain_params.h
Adds a new API for extracting codec film grain parameters as side data.
Adds a new AVFrameSideDataType entry AV_FRAME_DATA_FILM_GRAIN_PARAMS for it.
2020-10-28 - f95d9510ff - lavf 58.64.100 - avformat.h
Add AVSTREAM_EVENT_FLAG_NEW_PACKETS.
2020-09-28 - 68918d3b7f - lavu 56.60.100 - buffer.h
Add a av_buffer_replace() convenience function.
2020-09-13 - 837b6eb90e - lavu 56.59.100 - timecode.h
Add av_timecode_make_smpte_tc_string2.
2020-08-21 - 06f2651204 - lavu 56.58.100 - avstring.h
Deprecate av_d2str(). Use av_asprintf() instead.
2020-08-04 - 34de0abbe7 - lavu 56.58.100 - channel_layout.h
Add AV_CH_LAYOUT_22POINT2 together with its newly required pieces:
AV_CH_TOP_SIDE_LEFT, AV_CH_TOP_SIDE_RIGHT, AV_CH_BOTTOM_FRONT_CENTER,
AV_CH_BOTTOM_FRONT_LEFT, AV_CH_BOTTOM_FRONT_RIGHT.
2020-07-23 - 84655b7101 - lavu 56.57.100 - cpu.h
Add AV_CPU_FLAG_MMI and AV_CPU_FLAG_MSA.
2020-07-22 - 3a8e927176 - lavu 56.56.100 - imgutils.h
Add av_image_fill_plane_sizes().
2020-07-15 - 448a9aaa78 - lavc 58.96.100 - packet.h
Add AV_PKT_DATA_S12M_TIMECODE.
2020-06-12 - b09fb030c1 - lavu 56.55.100 - pixdesc.h
Add AV_PIX_FMT_X2RGB10.
2020-06-11 - bc8ab084fb - lavu 56.54.100 - frame.h
Add AV_FRAME_DATA_SEI_UNREGISTERED.
2020-06-10 - 1b4a98b029 - lavu 56.53.100 - log.h opt.h
Add av_opt_child_class_iterate() and AVClass.child_class_iterate().
Deprecate av_opt_child_class_next() and AVClass.child_class_next().
-------- 8< --------- FFmpeg 4.3 was cut here -------- 8< ---------
2020-06-05 - ec39c2276a - lavu 56.50.100 - buffer.h
Passing NULL as alloc argument to av_buffer_pool_init2() is now allowed.
2020-05-27 - ba6cada92e - lavc 58.88.100 - avcodec.h codec.h
Move AVCodec-related public API to new header codec.h.
2020-05-23 - 064b875e89 - lavu 56.49.100 - video_enc_params.h
Add AV_VIDEO_ENC_PARAMS_H264.
2020-05-23 - 2e08b39444 - lavu 56.48.100 - hwcontext.h
Add av_hwdevice_ctx_create_derived_opts.
2020-05-23 - 6b65c4ec54 - lavu 56.47.100 - rational.h
Add av_gcd_q().
2020-05-22 - af9e622776 - lavu 56.46.101 - opt.h
Add AV_OPT_FLAG_CHILD_CONSTS.
2020-05-22 - 9d443c3e68 - lavc 58.87.100 - avcodec.h codec_par.h
Move AVBitstreamFilter-related public API to new header bsf.h.
Move AVCodecParameters-related public API to new header codec_par.h.
2020-05-21 - 13b1bbff0b - lavc 58.86.101 - avcodec.h
Deprecated AV_CODEC_CAP_INTRA_ONLY and AV_CODEC_CAP_LOSSLESS.
2020-05-17 - 84af196c65 - lavu 56.46.100 - common.h
Add av_sat_add64() and av_sat_sub64()
2020-05-12 - 991d417692 - lavu 56.45.100 - video_enc_params.h
lavc 58.84.100 - avcodec.h
Add a new API for exporting video encoding information.
Replaces the deprecated API for exporting QP tables from decoders.
Add AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS to request this information from
decoders.
2020-05-10 - dccd07f66d - lavu 56.44.100 - hwcontext_vulkan.h
Add enabled_inst_extensions, num_enabled_inst_extensions, enabled_dev_extensions
and num_enabled_dev_extensions fields to AVVulkanDeviceContext
2020-04-22 - 0e1db79e37 - lavc 58.81.100 - packet.h
- lavu 56.43.100 - dovi_meta.h
Add AV_PKT_DATA_DOVI_CONF and AVDOVIDecoderConfigurationRecord.
2020-04-15 - 22b25b3ea5 - lavc 58.79.100 - avcodec.h
Add formal support for calling avcodec_flush_buffers() on encoders.
Encoders that set the cap AV_CODEC_CAP_ENCODER_FLUSH will be flushed.
For all other encoders, the call is now a no-op rather than undefined
behaviour.
2020-04-10 - 672946c7fe - lavc 58.78.100 - avcodec.h codec_desc.h codec_id.h packet.h
Move AVCodecDesc-related public API to new header codec_desc.h.
Move AVCodecID enum to new header codec_id.h.
Move AVPacket-related public API to new header packet.h.
2020-03-29 - 4cb0dda555 - lavf 58.42.100 - avformat.h
av_read_frame() now guarantees to handle uninitialized input packets
and to return refcounted packets on success.
2020-03-27 - c52ec0367d - lavc 58.77.100 - avcodec.h
av_packet_ref() now guarantees to return the destination packet
in a blank state on error.
2020-03-10 - 05d27f342b - lavc 58.75.100 - avcodec.h
Add AV_PKT_DATA_ICC_PROFILE.
2020-02-21 - d005a7cdfd - lavc 58.73.101 - avcodec.h
Add AV_CODEC_EXPORT_DATA_PRFT.
2020-02-21 - c666689491 - lavc 58.73.100 - avcodec.h
Add AVCodecContext.export_side_data and AV_CODEC_EXPORT_DATA_MVS.
2020-02-13 - e8f054b095 - lavu 56.41.100 - tx.h
Add AV_TX_INT32_FFT and AV_TX_INT32_MDCT
2020-02-12 - 3182114f88 - lavu 56.40.100 - log.h
Add av_log_once().
2020-02-04 - a88449ffb2 - lavu 56.39.100 - hwcontext.h
Add AV_PIX_FMT_VULKAN
Add AV_HWDEVICE_TYPE_VULKAN and implementation.
2020-01-30 - 27529eeb27 - lavf 58.37.100 - avio.h
Add avio_protocol_get_class().
2020-01-15 - 717b2074ec - lavc 58.66.100 - avcodec.h
Add AV_PKT_DATA_PRFT and AVProducerReferenceTime.
2019-12-27 - 45259a0ee4 - lavu 56.38.100 - eval.h
Add av_expr_count_func().
2019-12-26 - 16685114d5 - lavu 56.37.100 - buffer.h
Add av_buffer_pool_buffer_get_opaque().
2019-11-17 - 1c23abc88f - lavu 56.36.100 - eval API
Add av_expr_count_vars().
2019-10-14 - f3746d31f9 - lavu 56.35.101 - opt.h
Add AV_OPT_FLAG_RUNTIME_PARAM.
2019-09-25 - f8406ab4b9 - lavc 58.59.100 - avcodec.h
Add max_samples
2019-09-04 - 2a9d461abc - lavu 56.35.100 - hwcontext_videotoolbox.h
Add av_map_videotoolbox_format_from_pixfmt2() for full range pixfmt
2019-09-01 - 8821d1f56e - lavu 56.34.100 - pixfmt.h
Add EBU Tech. 3213-E AVColorPrimaries value
2019-08-17 - 95fa73a2b4 - lavf 58.31.101 - avio.h
4K limit removed from avio_printf.
2019-08-17 - a82f8f2f10 - lavf 58.31.100 - avio.h
Add avio_print_string_array and avio_print.
2019-07-27 - 42e2319ba9 - lavu 56.33.100 - tx.h
Add AV_TX_DOUBLE_FFT and AV_TX_DOUBLE_MDCT
-------- 8< --------- FFmpeg 4.2 was cut here -------- 8< ---------
2019-06-21 - a30e44098a - lavu 56.30.100 - frame.h
Add FF_DECODE_ERROR_DECODE_SLICES
2019-06-14 - edfced8c04 - lavu 56.29.100 - frame.h
Add FF_DECODE_ERROR_CONCEALMENT_ACTIVE
2019-05-15 - b79b29ddb1 - lavu 56.28.100 - tx.h
Add av_tx_init(), av_tx_uninit() and related definitions.
2019-04-20 - 3153a6502a - lavc 58.52.100 - avcodec.h
Add AV_CODEC_FLAG_DROPCHANGED to allow avcodec_receive_frame to drop
frames whose parameters differ from first decoded frame in stream.
2019-04-12 - abfeba9724 - lavf 58.27.102
Rename hls,applehttp demuxer to hls
2019-01-27 - 5bcefceec8 - lavc 58.46.100 - avcodec.h
Add discard_damaged_percentage
2019-01-08 - 1ef4828276 - lavu 56.26.100 - frame.h
Add AV_FRAME_DATA_REGIONS_OF_INTEREST
2018-12-21 - 2744d6b364 - lavu 56.25.100 - hdr_dynamic_metadata.h
Add AV_FRAME_DATA_DYNAMIC_HDR_PLUS enum value, av_dynamic_hdr_plus_alloc(),
av_dynamic_hdr_plus_create_side_data() functions, and related structs.
-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
Add av_read_image_line2(), av_write_image_line2()
2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
Add AV_FRAME_DATA_S12M_TIMECODE
2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
Add av_mediacodec_render_buffer_at_time().
2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
Add AV_PKT_DATA_AFD
2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
Add av_bsf_flush().
2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
Add pmt_version field to AVProgram
2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
Add AV_DISPOSITION_STILL_IMAGE
2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
Add AVCUDADeviceContext.stream.
2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
Add AV_OPT_FLAG_DEPRECATED.
2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
Add av_thread_message_queue_nb_elems().
-------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
Deprecate AV_PIX_FMT_FLAG_PSEUDOPAL and make allocating a pseudo palette
optional for API users (see AV_PIX_FMT_FLAG_PSEUDOPAL doxygen for details).
2018-04-01 - 860086ee16 - lavc 58.17.100 - avcodec.h
Add av_packet_make_refcounted().
2018-04-01 - f1805d160d - lavfi 7.14.100 - avfilter.h
Deprecate use of avfilter_register(), avfilter_register_all(),
avfilter_next(). Add av_filter_iterate().
2018-03-25 - b7d0d912ef - lavc 58.16.100 - avcodec.h
Add FF_SUB_CHARENC_MODE_IGNORE.
2018-03-23 - db2a7c947e - lavu 56.12.100 - encryption_info.h
Add AVEncryptionInitInfo and AVEncryptionInfo structures to hold new side-data
for encryption info.
2018-03-21 - f14ca60001 - lavc 58.15.100 - avcodec.h
Add av_packet_make_writable().
2018-03-18 - 4b86ac27a0 - lavu 56.11.100 - frame.h
Add AV_FRAME_DATA_QP_TABLE_PROPERTIES and AV_FRAME_DATA_QP_TABLE_DATA.
2018-03-15 - e0e72539cf - lavu 56.10.100 - opt.h
Add AV_OPT_FLAG_BSF_PARAM
2018-03-07 - 950170bd3b - lavu 56.9.100 - crc.h
Add AV_CRC_8_EBU crc variant.
2018-03-07 - 2a0eb86857 - lavc 58.14.100 - mediacodec.h
Change the default behavior of avcodec_flush() on mediacodec
video decoders. To restore the previous behavior, use the new
delay_flush=1 option.
2018-03-01 - 6731f60598 - lavu 56.8.100 - frame.h
Add av_frame_new_side_data_from_buf().
2018-02-15 - 8a8d0b319a
Change av_ripemd_update(), av_murmur3_update() and av_hash_update() length
parameter type to size_t at next major bump.
2018-02-12 - bcab11a1a2 - lavfi 7.12.100 - avfilter.h
Add AVFilterContext.extra_hw_frames.
2018-02-12 - d23fff0d8a - lavc 58.11.100 - avcodec.h
Add AVCodecContext.extra_hw_frames.
2018-02-06 - 0694d87024 - lavf 58.9.100 - avformat.h
Deprecate use of av_register_input_format(), av_register_output_format(),
av_register_all(), av_iformat_next(), av_oformat_next().
Add av_demuxer_iterate(), and av_muxer_iterate().
2018-02-06 - 36c85d6e77 - lavc 58.10.100 - avcodec.h
Deprecate use of avcodec_register(), avcodec_register_all(),
av_codec_next(), av_register_codec_parser(), and av_parser_next().
Add av_codec_iterate() and av_parser_iterate().
2018-02-04 - ff46124b0d - lavf 58.8.100 - avformat.h
Deprecate the current names of the RTSP "timeout", "stimeout", "user-agent"
options. Introduce "listen_timeout" as replacement for the current "timeout"
option, and "user_agent" as replacement for "user-agent". Once the deprecation
is over, the old "timeout" option will be removed, and "stimeout" will be
renamed to "stimeout" (the "timeout" option will essentially change semantics).
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Deprecate AVFormatContext filename field which had limited length, use the
new dynamically allocated url field instead.
2018-01-28 - ea3672b7d6 - lavf 58.7.100 - avformat.h
Add url field to AVFormatContext and add ff_format_set_url helper function.
2018-01-27 - 6194d7e564 - lavf 58.6.100 - avformat.h
Add AVFMTCTX_UNSEEKABLE (for HLS demuxer).
2018-01-23 - 9f07cf7c00 - lavu 56.9.100 - aes_ctr.h
Add method to set the 16-byte IV.
2018-01-16 - 631c56a8e4 - lavf 58.5.100 - avformat.h
Explicitly make avformat_network_init() and avformat_network_deinit() optional.
If these are not called, network initialization and deinitialization is
automatic, and unlike in older versions, fully supported, unless libavformat
is linked to ancient GnuTLS and OpenSSL.
2018-01-16 - 6512ff72f9 - lavf 58.4.100 - avformat.h
Deprecate AVStream.recommended_encoder_configuration. It was useful only for
FFserver, which has been removed.
2018-01-05 - 798dcf2432 - lavfi 7.11.101 - avfilter.h
Deprecate avfilter_link_get_channels(). Use av_buffersink_get_channels().
2017-01-04 - c29038f304 - lavr 4.0.0 - avresample.h
Deprecate the entire library. Merged years ago to provide compatibility
with Libav, it remained unmaintained by the FFmpeg project and duplicated
functionality provided by libswresample.
In order to improve consistency and reduce attack surface, it has been deprecated.
Users of this library are asked to migrate to libswresample, which, as well as
providing more functionality, is faster and has higher accuracy.
2017-12-26 - a04c2c707d - lavc 58.9.100 - avcodec.h
Deprecate av_lockmgr_register(). You need to build FFmpeg with threading
support enabled to get basic thread-safety (which is the default build
configuration).
2017-12-24 - 8b81eabe57 - lavu 56.7.100 - cpu.h
AVX-512 flags added.
2017-12-16 - 8bf4e6d3ce - lavc 58.8.100 - avcodec.h
The MediaCodec decoders now support AVCodecContext.hw_device_ctx.
2017-12-16 - e4d9f05ca7 - lavu 56.6.100 - hwcontext.h hwcontext_mediacodec.h
Add AV_HWDEVICE_TYPE_MEDIACODEC and a new installed header with
MediaCodec-specific hwcontext definitions.
2017-12-14 - b945fed629 - lavc 58.7.100 - avcodec.h
Add AV_CODEC_CAP_HARDWARE, AV_CODEC_CAP_HYBRID, and AVCodec.wrapper_name,
and mark all AVCodecs accordingly.
2017-11-29 - d268094f88 - lavu 56.4.100 / 56.7.0 - stereo3d.h
Add view field to AVStereo3D structure and AVStereo3DView enum.
2017-11-26 - 3a71bcc213 - lavc 58.6.100 - avcodec.h
Add const to AVCodecContext.hwaccel.
2017-11-26 - 3536a3efb9 - lavc 58.5.100 - avcodec.h
Deprecate user visibility of the AVHWAccel structure and the functions
av_register_hwaccel() and av_hwaccel_next().
2017-11-26 - 24cc0a53e9 - lavc 58.4.100 - avcodec.h
Add AVCodecHWConfig and avcodec_get_hw_config().
2017-11-22 - 3650cb2dfa - lavu 56.3.100 - opencl.h
Remove experimental OpenCL API (av_opencl_*).
2017-11-22 - b25d8ef0a7 - lavu 56.2.100 - hwcontext.h hwcontext_opencl.h
Add AV_HWDEVICE_TYPE_OPENCL and a new installed header with
OpenCL-specific hwcontext definitions.
2017-11-22 - a050f56c09 - lavu 56.1.100 - pixfmt.h
Add AV_PIX_FMT_OPENCL.
2017-11-11 - 48e4eda11d - lavc 58.3.100 - avcodec.h
Add avcodec_get_hw_frames_parameters().
-------- 8< --------- FFmpeg 3.4 was cut here -------- 8< ---------
2017-09-28 - b6cf66ae1c - lavc 57.106.104 - avcodec.h
@@ -1197,7 +742,7 @@ API changes, most recent first:
Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
dictionary types being set as options.
2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
the user when events happen in the file/stream.
@@ -1214,7 +759,7 @@ API changes, most recent first:
2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
Add avio_feof() and deprecate url_feof().
2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
sws_getContext is not going to be removed in the future.
2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h

View File

@@ -38,7 +38,7 @@ PROJECT_NAME = FFmpeg
# could be handy for archiving the generated documentation or if some version
# control system is used.
PROJECT_NUMBER =
PROJECT_NUMBER = 3.4.1
# Using the PROJECT_BRIEF tag one can provide an optional one line description
# for a project that appears at the top of each page and should give viewer a

View File

@@ -37,61 +37,6 @@ raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
to MOV/MP4 files and related formats such as 3GP or M4A. Please note
that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
@section av1_metadata
Modify metadata embedded in an AV1 stream.
@table @option
@item td
Insert or remove temporal delimiter OBUs in all temporal units of the
stream.
@table @samp
@item insert
Insert a TD at the beginning of every TU which does not already have one.
@item remove
Remove the TD from the beginning of every TU which has one.
@end table
@item color_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).
@item color_range
Set the color range in the stream (see AV1 section 6.4.2; note that
this cannot be set for streams using BT.709 primaries, sRGB transfer
characteristic and identity (RGB) matrix coefficients).
@table @samp
@item tv
Limited range.
@item pc
Full range.
@end table
@item chroma_sample_position
Set the chroma sample location in the stream (see AV1 section 6.4.2).
This can only be set for 4:2:0 streams.
@table @samp
@item vertical
Left position (matching the default in MPEG-2 and H.264).
@item colocated
Top-left position.
@end table
@item tick_rate
Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
the timing info in the sequence header.
@item num_ticks_per_picture
Set the number of ticks in each picture, to indicate that the stream
has a fixed framerate. Ignored if @option{tick_rate} is not also set.
@item delete_padding
Deletes Padding OBUs.
@end table
@section chomp
Remove zero padding at the end of a packet.
@@ -103,24 +48,21 @@ DTS-HD.
@section dump_extra
Add extradata to the beginning of the filtered packets except when
said packets already exactly begin with the extradata that is intended
to be added.
Add extradata to the beginning of the filtered packets.
@table @option
@item freq
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item a
add extradata to all key packets, but only if @var{local_header} is
set in the @option{flags2} codec context field
@item k
@item keyframe
add extradata to all key packets
@item e
@item all
add extradata to all packets
@end table
@end table
If not specified it is assumed @samp{k}.
@@ -132,10 +74,6 @@ the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@section extract_extradata
Extract the in-band extradata.
@@ -154,139 +92,6 @@ When this option is enabled, the long-term headers are removed from the
bitstream after extraction.
@end table
@section filter_units
Remove units with types in or not in a given set from the stream.
@table @option
@item pass_types
List of unit types or ranges of unit types to pass through while removing
all others. This is specified as a '|'-separated list of unit type values
or ranges of values with '-'.
@item remove_types
Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
Extradata is unchanged by this transformation, but note that if the stream
contains inline parameter sets then the output may be unusable if they are
removed.
For example, to remove all non-VCL NAL units from an H.264 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
@end example
To remove all AUDs, SEI and filler from an H.265 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
@section hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
@table @option
@item texture
Specifies the texture to keep.
@table @option
@item color
@item alpha
@end table
@end table
Convert HAPQA to HAPQ
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
@end example
Convert HAPQA to HAPAlphaOnly
@example
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
@end example
@section h264_metadata
Modify metadata embedded in an H.264 stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item overscan_appropriate_flag
Set whether the stream is suitable for display using overscan
or not (see H.264 section E.2.1).
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and
table E-2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.264 section E.2.1
and tables E-3, E-4 and E-5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section
E.2.1 and figure E-1).
@item tick_rate
Set the tick rate (num_units_in_tick / time_scale) in the VUI
parameters. This is the smallest time unit representable in the
stream, and in many cases represents the field rate of the stream
(double the frame rate).
@item fixed_frame_rate_flag
Set whether the stream has fixed framerate - typically this indicates
that the framerate is exactly half the tick rate, but the exact
meaning is dependent on interlacing and the picture structure (see
H.264 section E.2.1 and table E-6).
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the frame cropping offsets in the SPS. These values will replace
the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled or the stream is interlaced
(see H.264 section 7.4.2.1.1).
@item sei_user_data
Insert a string as SEI unregistered user data. The argument must
be of the form @emph{UUID+string}, where the UUID is as hex digits
possibly separated by hyphens, and the string can be anything.
For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1
to A-5.
The argument must be the name of a level (for example, @samp{4.2}), a
level_idc value (for example, @samp{42}), or the special name @samp{auto}
indicating that the filter should attempt to guess the level from the
input stream properties.
@end table
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
@@ -306,78 +111,6 @@ ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
Modify metadata embedded in an HEVC stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and
table E.2).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.265 section E.3.1
and tables E.3, E.4 and E.5).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section
E.3.1 and figure E.1).
@item tick_rate
Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
time_scale). Combined with @option{num_ticks_poc_diff_one}, this can
set a constant framerate in the stream. Note that it is likely to be
overridden by container parameters when the stream is in a container.
@item num_ticks_poc_diff_one
Set poc_proportional_to_timing_flag in VPS and VUI and use this value
to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and
E.3.1). Ignored if @option{tick_rate} is not also set.
@item crop_left
@item crop_right
@item crop_top
@item crop_bottom
Set the conformance window cropping offsets in the SPS. These values
will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be
representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
@item level
Set the level in the VPS and SPS. See H.265 section A.4 and tables
A.6 and A.7.
The argument must be the name of a level (for example, @samp{5.1}), a
@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
or the special name @samp{auto} indicating that the filter should
attempt to guess the level from the input stream properties.
@end table
@section hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code
@@ -465,42 +198,6 @@ See also the @ref{text2movsub} filter.
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@table @option
@item display_aspect_ratio
Set the display aspect ratio in the stream.
The following fixed values are supported:
@table @option
@item 4/3
@item 16/9
@item 221/100
@end table
Any other value will result in square pixels being signalled instead
(see H.262 section 6.3.3 and table 6-3).
@item frame_rate
Set the frame rate in the stream. This is constructed from a table
of known values combined with a small multiplier and divisor - if
the supplied value is not exactly representable, the nearest
representable value will be used instead (see H.262 section 6.3.3
and table 6-4).
@item video_format
Set the video format in the stream (see H.262 section 6.3.6 and
table 6-6).
@item colour_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the colour description in the stream (see H.262 section 6.3.6
and tables 6-7, 6-8 and 6-9).
@end table
@section mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
@@ -548,111 +245,6 @@ ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@section null
This bitstream filter passes the packets through unchanged.
@section pcm_rechunk
Repacketize PCM audio to a fixed number of samples per packet or a fixed packet
rate per second. This is similar to the @ref{asetnsamples,,asetnsamples audio
filter,ffmpeg-filters} but works on audio packets instead of audio frames.
@table @option
@item nb_out_samples, n
Set the number of samples per each output audio packet. The number is intended
as the number of samples @emph{per each channel}. Default value is 1024.
@item pad, p
If set to 1, the filter will pad the last audio packet with silence, so that it
will contain the same number of samples (or roughly the same number of samples,
see @option{frame_rate}) as the previous ones. Default value is 1.
@item frame_rate, r
This option makes the filter output a fixed number of packets per second instead
of a fixed number of samples per packet. If the audio sample rate is not
divisible by the frame rate then the number of samples will not be constant but
will vary slightly so that each packet will start as close to the frame
boundary as possible. Using this option has precedence over @option{nb_out_samples}.
@end table
You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
for NTSC frame rate using the @option{frame_rate} option.
@example
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section prores_metadata
Modify color property metadata embedded in prores stream.
@table @option
@item color_primaries
Set the color primaries.
Available values are:
@table @samp
@item auto
Keep the same color primaries property (default).
@item unknown
@item bt709
@item bt470bg
BT601 625
@item smpte170m
BT601 525
@item bt2020
@item smpte431
DCI P3
@item smpte432
P3 D65
@end table
@item transfer_characteristics
Set the color transfer.
Available values are:
@table @samp
@item auto
Keep the same transfer characteristics property (default).
@item unknown
@item bt709
BT 601, BT 709, BT 2020
@item smpte2084
SMPTE ST 2084
@item arib-std-b67
ARIB STD-B67
@end table
@item matrix_coefficients
Set the matrix coefficient.
Available values are:
@table @samp
@item auto
Keep the same colorspace property (default).
@item unknown
@item bt709
@item smpte170m
BT 601
@item bt2020nc
@end table
@end table
Set Rec709 colorspace for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
@end example
Set Hybrid Log-Gamma parameters for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
@end example
@section remove_extra
Remove extradata from packets.
@@ -675,63 +267,6 @@ Remove extradata from all frames.
@end table
@end table
@section setts
Set PTS and DTS in packets.
It accepts the following parameters:
@table @option
@item ts
@item pts
@item dts
Set expressions for PTS, DTS or both.
@end table
The expressions are evaluated through the eval API and can contain the following
constants:
@table @option
@item N
The count of the input packet. Starting from 0.
@item TS
The demux timestamp in input in case of @code{ts} or @code{dts} option or presentation
timestamp in case of @code{pts} option.
@item POS
The original position in the file of the packet, or undefined if undefined
for the current packet
@item DTS
The demux timestamp in input.
@item PTS
The presentation timestamp in input.
@item STARTDTS
The DTS of the first packet.
@item STARTPTS
The PTS of the first packet.
@item PREV_INDTS
The previous input DTS.
@item PREV_INPTS
The previous input PTS.
@item PREV_OUTDTS
The previous output DTS.
@item PREV_OUTPTS
The previous output PTS.
@item TB
The timebase of stream packet belongs.
@item SR
The sample rate of stream packet belongs.
@end table
@anchor{text2movsub}
@section text2movsub
@@ -740,47 +275,6 @@ codec) with metadata headers.
See also the @ref{mov2textsub} filter.
@section trace_headers
Log trace output containing all syntax elements in the coded stream
headers (everything above the level of individual coded blocks).
This can be useful for debugging low-level stream issues.
Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending
on the build only a subset of these may be available.
@section truehd_core
Extract the core from a TrueHD stream, dropping ATMOS data.
@section vp9_metadata
Modify metadata embedded in a VP9 stream.
@table @option
@item color_space
Set the color space value in the frame header. Note that any frame
set to RGB will be implicitly set to PC range and that RGB is
incompatible with profiles 0 and 2.
@table @samp
@item unknown
@item bt601
@item bt709
@item smpte170
@item smpte240
@item bt2020
@item rgb
@end table
@item color_range
Set the color range value in the frame header. Note that any value
imposed by the color space will take precedence over this value.
@table @samp
@item tv
@item pc
@end table
@end table
@section vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This

View File

@@ -36,11 +36,11 @@ install
examples
Build all examples located in doc/examples.
checkheaders
Check headers dependencies.
libavformat/output-example
Build the libavformat basic example.
alltools
Build all tools in tools directory.
libswscale/swscale-test
Build the swscale self-test (useful also as an example).
config
Reconfigure the project with the current configuration.
@@ -48,8 +48,6 @@ config
tools/target_dec_<decoder>_fuzzer
Build fuzzer to fuzz the specified decoder.
tools/target_bsf_<filter>_fuzzer
Build fuzzer to fuzz the specified bitstream filter.
Useful standard make commands:
make -t <target>

View File

@@ -44,20 +44,26 @@ Use 1/4 pel motion compensation.
Use loop filter.
@item qscale
Use fixed qscale.
@item gmc
Use gmc.
@item mv0
Always try a mb with mv=<0,0>.
@item input_preserved
@item pass1
Use internal 2pass ratecontrol in first pass mode.
@item pass2
Use internal 2pass ratecontrol in second pass mode.
@item gray
Only decode/encode grayscale.
@item emu_edge
Do not draw edges.
@item psnr
Set error[?] variables during encoding.
@item truncated
Input bitstream might be randomly truncated.
@item drop_changed
Don't output frames whose parameters differ from first decoded frame in stream.
Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
@item naq
Normalize adaptive quantization.
@item ildct
Use interlaced DCT.
@item low_delay
@@ -70,14 +76,50 @@ This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item aic
Apply H263 advanced intra coding / mpeg4 ac prediction.
@item cbp
Deprecated, use mpegvideo private options instead.
@item qprd
Deprecated, use mpegvideo private options instead.
@item ilme
Apply interlaced motion estimation.
@item cgop
Use closed gop.
@item output_corrupt
Output even potentially corrupted frames.
@end table
@item me_method @var{integer} (@emph{encoding,video})
Set motion estimation method.
Possible values:
@table @samp
@item zero
zero motion estimation (fastest)
@item full
full motion estimation (slowest)
@item epzs
EPZS motion estimation (default)
@item esa
esa motion estimation (alias for full)
@item tesa
tesa motion estimation
@item dia
dia motion estimation (alias for epzs)
@item log
log motion estimation
@item phods
phods motion estimation
@item x1
X1 motion estimation
@item hex
hex motion estimation
@item umh
umh motion estimation
@item iter
iter motion estimation
@end table
@item extradata_size @var{integer}
Set extradata size.
@item time_base @var{rational number}
Set codec time base.
@@ -144,6 +186,9 @@ Default value is 0.
@item b_qfactor @var{float} (@emph{encoding,video})
Set qp factor between P and B frames.
@item rc_strategy @var{integer} (@emph{encoding,video})
Set ratecontrol method.
@item b_strategy @var{integer} (@emph{encoding,video})
Set strategy to choose between I/P/B-frames.
@@ -167,6 +212,8 @@ Possible values:
@table @samp
@item autodetect
@item old_msmpeg4
some old lavc generated msmpeg4v3 files (no autodetection)
@item xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)
@item ump4
@@ -175,6 +222,8 @@ Xvid interlacing bug (autodetected if fourcc==XVIX)
padding bug (autodetected)
@item amv
@item ac_vlc
illegal vlc bug (autodetected per fourcc)
@item qpel_chroma
@item std_qpel
@@ -195,6 +244,14 @@ Workaround various bugs in microsoft broken decoders.
trancated frames
@end table
@item lelim @var{integer} (@emph{encoding,video})
Set single coefficient elimination threshold for luminance (negative
values also consider DC coefficient).
@item celim @var{integer} (@emph{encoding,video})
Set single coefficient elimination threshold for chrominance (negative
values also consider dc coefficient)
@item strict @var{integer} (@emph{decoding/encoding,audio,video})
Specify how strictly to follow the standards.
@@ -251,8 +308,26 @@ consider things that a sane encoder should not do as an error
@item mpeg_quant @var{integer} (@emph{encoding,video})
Use MPEG quantizers instead of H.263.
@item qsquish @var{float} (@emph{encoding,video})
How to keep quantizer between qmin and qmax (0 = clip, 1 = use
differentiable function).
@item rc_qmod_amp @var{float} (@emph{encoding,video})
Set experimental quantizer modulation.
@item rc_qmod_freq @var{integer} (@emph{encoding,video})
Set experimental quantizer modulation.
@item rc_override_count @var{integer}
@item rc_eq @var{string} (@emph{encoding,video})
Set rate control equation. When computing the expression, besides the
standard functions defined in the section 'Expression Evaluation', the
following functions are available: bits2qp(bits), qp2bits(qp). Also
the following constants are available: iTex pTex tex mv fCode iCount
mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex
avgTex.
@item maxrate @var{integer} (@emph{encoding,audio,video})
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
@@ -263,12 +338,18 @@ encode. It is of little use elsewise.
@item bufsize @var{integer} (@emph{encoding,audio,video})
Set ratecontrol buffer size (in bits).
@item rc_buf_aggressivity @var{float} (@emph{encoding,video})
Currently useless.
@item i_qfactor @var{float} (@emph{encoding,video})
Set QP factor between P and I frames.
@item i_qoffset @var{float} (@emph{encoding,video})
Set QP offset between P and I frames.
@item rc_init_cplx @var{float} (@emph{encoding,video})
Set initial complexity for 1-pass encoding.
@item dct @var{integer} (@emph{encoding,video})
Set DCT algorithm.
@@ -333,7 +414,11 @@ Automatically pick a IDCT compatible with the simple one
@item simpleneon
@item xvid
@item simplealpha
@item ipp
@item xvidmmx
@item faani
floating point AAN IDCT
@@ -390,6 +475,8 @@ rate control
macroblock (MB) type
@item qp
per-block quantization parameter (QP)
@item mv
motion vector
@item dct_coeff
@item green_metadata
@@ -399,12 +486,18 @@ display complexity metadata for the upcoming frame, GoP or for a given duration.
@item startcode
@item pts
@item er
error recognition
@item mmco
memory management control operations (H.264)
@item bugs
@item vis_qp
visualize quantization parameter (QP), lower QP are tinted greener
@item vis_mb_type
visualize block types
@item buffers
picture buffer allocations
@item thread_ops
@@ -413,6 +506,21 @@ threading operations
skip motion compensation
@end table
@item vismv @var{integer} (@emph{decoding,video})
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
@table @samp
@item pf
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@item cmp @var{integer} (@emph{encoding,video})
Set full pel me compare function.
@@ -563,24 +671,6 @@ noise preserving sum of squared differences
@item dia_size @var{integer} (@emph{encoding,video})
Set diamond type & size for motion estimation.
@table @samp
@item (1024, INT_MAX)
full motion estimation(slowest)
@item (768, 1024]
umh motion estimation
@item (512, 768]
hex motion estimation
@item (256, 512]
l2s diamond motion estimation
@item [2,256]
var diamond motion estimation
@item (-1, 2)
small diamond motion estimation
@item -1
funny diamond motion estimation
@item (INT_MIN, -1)
sab diamond motion estimation
@end table
@item last_pred @var{integer} (@emph{encoding,video})
Set amount of motion predictors from the previous frame.
@@ -631,9 +721,19 @@ Set diamond type & size for motion estimation pre-pass.
@item subq @var{integer} (@emph{encoding,video})
Set sub pel motion estimation quality.
@item dtg_active_format @var{integer}
@item me_range @var{integer} (@emph{encoding,video})
Set limit motion vectors range (1023 for DivX player).
@item ibias @var{integer} (@emph{encoding,video})
Set intra quant bias.
@item pbias @var{integer} (@emph{encoding,video})
Set inter quant bias.
@item color_table_id @var{integer}
@item global_quality @var{integer} (@emph{encoding,audio,video})
@item coder @var{integer} (@emph{encoding,video})
@@ -648,6 +748,8 @@ arithmetic coder
raw (no encoding)
@item rle
run-length coder
@item deflate
deflate-based coder
@end table
@item context @var{integer} (@emph{encoding,video})
@@ -655,6 +757,8 @@ Set context model.
@item slice_flags @var{integer}
@item xvmc_acceleration @var{integer}
@item mbd @var{integer} (@emph{encoding,video})
Set macroblock decision algorithm (high quality mode).
@@ -668,9 +772,17 @@ use fewest bits
use best rate distortion
@end table
@item stream_codec_tag @var{integer}
@item sc_threshold @var{integer} (@emph{encoding,video})
Set scene change threshold.
@item lmin @var{integer} (@emph{encoding,video})
Set min lagrange factor (VBR).
@item lmax @var{integer} (@emph{encoding,video})
Set max lagrange factor (VBR).
@item nr @var{integer} (@emph{encoding,video})
Set noise reduction.
@@ -678,12 +790,14 @@ Set noise reduction.
Set number of bits which should be loaded into the rc buffer before
decoding starts.
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
Possible values:
@table @samp
@item fast
Allow non spec compliant speedup tricks.
@item sgop
Deprecated, use mpegvideo private options instead.
@item noout
Skip bitstream encoding.
@item ignorecrop
@@ -694,32 +808,17 @@ Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
@item showall
Show all frames before the first keyframe.
@item skiprd
Deprecated, use mpegvideo private options instead.
@item export_mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item skip_manual
Do not skip samples and export skip information as frame side data.
@item ass_ro_flush_noop
Do not reset ASS ReadOrder field on flush.
@end table
@item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})
@item error @var{integer} (@emph{encoding,video})
Possible values:
@table @samp
@item mvs
Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
for codecs that support it. See also @file{doc/examples/export_mvs.c}.
@item prft
Export encoder Producer Reference Time into packet side-data (see @code{AV_PKT_DATA_PRFT})
for codecs that support it.
@item venc_params
Export video encoding parameters through frame side data (see @code{AV_FRAME_DATA_VIDEO_ENC_PARAMS})
for codecs that support it. At present, those are H.264 and VP9.
@item film_grain
Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
Supported at present by AV1 decoders.
@end table
@item qns @var{integer} (@emph{encoding,video})
Deprecated, use mpegvideo private options instead.
@item threads @var{integer} (@emph{decoding/encoding,video})
Set the number of threads to be used, in case the selected codec
@@ -733,6 +832,12 @@ automatically select the number of threads to set
Default value is @samp{auto}.
@item me_threshold @var{integer} (@emph{encoding,video})
Set motion estimation threshold.
@item mb_threshold @var{integer} (@emph{encoding,video})
Set macroblock threshold.
@item dc @var{integer} (@emph{encoding,video})
Set intra_dc_precision.
@@ -747,8 +852,49 @@ Set number of macroblock rows at the bottom which are skipped.
@item profile @var{integer} (@emph{encoding,audio,video})
Set encoder codec profile. Default value is @samp{unknown}. Encoder specific
profiles are documented in the relevant encoder documentation.
Possible values:
@table @samp
@item unknown
@item aac_main
@item aac_low
@item aac_ssr
@item aac_ltp
@item aac_he
@item aac_he_v2
@item aac_ld
@item aac_eld
@item mpeg2_aac_low
@item mpeg2_aac_he
@item mpeg4_sp
@item mpeg4_core
@item mpeg4_main
@item mpeg4_asp
@item dts
@item dts_es
@item dts_96_24
@item dts_hd_hra
@item dts_hd_ma
@end table
@item level @var{integer} (@emph{encoding,audio,video})
@@ -810,6 +956,9 @@ noise preserving sum of squared differences
@end table
@item border_mask @var{float} (@emph{encoding,video})
Increase the quantizer for macroblocks close to borders.
@item mblmin @var{integer} (@emph{encoding,video})
Set min macroblock lagrange factor (VBR).
@@ -846,9 +995,6 @@ Discard all bidirectional frames.
@item nokey
Discard all frames excepts keyframes.
@item nointra
Discard all frames except I frames.
@item all
Discard all frames.
@end table
@@ -873,6 +1019,10 @@ Set chroma qp offset from luma.
@item trellis @var{integer} (@emph{encoding,audio,video})
Set rate-distortion optimal quantization.
@item sc_factor @var{integer} (@emph{encoding,video})
Set value multiplied by qscale for each frame and added to
scene_change_score.
@item mv0_threshold @var{integer} (@emph{encoding,video})
@item b_sensitivity @var{integer} (@emph{encoding,video})
Adjust sensitivity of b_frame_strategy 1.
@@ -883,6 +1033,9 @@ Adjust sensitivity of b_frame_strategy 1.
@item timecode_frame_start @var{integer} (@emph{encoding,video})
Set GOP timecode frame start number, in non drop frame format.
@item request_channels @var{integer} (@emph{decoding,audio})
Set desired number of audio channels.
@item bits_per_raw_sample @var{integer}
@item channel_layout @var{integer} (@emph{decoding/encoding,audio})
@@ -996,12 +1149,6 @@ BT.2020 NCL
BT.2020 CL
@item smpte2085
SMPTE 2085
@item chroma-derived-nc
Chroma-derived NCL
@item chroma-derived-c
Chroma-derived CL
@item ictcp
ICtCp
@end table
@item color_range @var{integer} (@emph{decoding/encoding,video})
@@ -1122,7 +1269,7 @@ instead of alpha. Default is 0.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg

View File

@@ -25,18 +25,12 @@ enabled decoders.
A description of some of the currently available video decoders
follows.
@section av1
@section hevc
AOMedia Video 1 (AV1) decoder.
HEVC / H.265 decoder.
@subsection Options
@table @option
@item operating_point
Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.
@end table
Note: the @option{skip_loop_filter} option has effect only at level
@code{all}.
@section rawvideo
@@ -60,66 +54,6 @@ top-field-first is assumed
@end table
@section libdav1d
dav1d AV1 decoder.
libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
Requires the presence of the libdav1d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libdav1d}.
@subsection Options
The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
@item filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the
internal default of the library.
@item oppoint
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
internal default of the library.
@item alllayers
Output all spatial layers of a scalable AV1 bitstream. The default value is false.
@end table
@section libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.
This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@section libuavs3d
AVS3-P2/IEEE1857.10 video decoder.
libuavs3d allows libavcodec to decode AVS3 streams.
Requires the presence of the libuavs3d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libuavs3d}.
@subsection Options
The following option is supported by the libuavs3d wrapper.
@table @option
@item frame_threads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@@ -141,7 +75,7 @@ the undocumented RealAudio 3 (a.k.a. dnet).
@item -drc_scale @var{value}
Dynamic Range Scale Factor. The factor to apply to dynamic range values
from the AC-3 stream. This factor is applied exponentially. The default value is 1.
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
@@ -261,31 +195,6 @@ without this library.
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section libaribb24
ARIB STD-B24 caption decoder.
Implements profiles A and C of the ARIB STD-B24 standard.
@subsection libaribb24 Decoder Options
@table @option
@item -aribb24-base-path @var{path}
Sets the base path for the libaribb24 library. This is utilized for reading of
configuration files (for custom unicode conversions), and for dumping of
non-text symbols as images under that location.
Unset by default.
@item -aribb24-skip-ruby-text @var{boolean}
Tells the decoder wrapper to skip text blocks that contain half-height ruby
text.
Enabled by default.
@end table
@section dvbsub
@subsection Options
@@ -321,7 +230,7 @@ palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@@ -346,30 +255,18 @@ configuration. You need to explicitly configure the build with
@table @option
@item txt_page
List of teletext page numbers to decode. Pages that do not match the specified
list are dropped. You may use the special @code{*} string to match all pages,
or @code{subtitle} to match all subtitle pages.
List of teletext page numbers to decode. You may use the special * string to
match all pages. Pages that do not match the specified list are dropped.
Default value is *.
@item txt_default_region
Set default character set used for decoding, a value between 0 and 87 (see
ETS 300 706, Section 15, Table 32). Default value is -1, which does not
override the libzvbi default. This option is needed for some legacy level 1.0
transmissions which cannot signal the proper charset.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles.
@table @option
@item bitmap
The default format, you should use this for teletext pages, because certain
graphics and colors cannot be expressed in simple text or even ASS.
@item text
Simple text based output without formatting.
@item ass
Formatted ASS output, subtitle pages and teletext pages are returned in
different styles, subtitle pages are stripped down to text, but an effort is
made to keep the text alignment and the formatting.
@end table
Specifies the format of the decoded subtitles. The teletext decoder is capable
of decoding the teletext pages to bitmaps or to simple text, you should use
"bitmap" for teletext pages, because certain graphics and colors cannot be
expressed in simple text. You might use "text" for teletext based subtitles if
your application can handle simple text based subtitles. Default value is
bitmap.
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
@@ -382,8 +279,7 @@ present between the subtitle lines because of double-sized teletext characters.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. Default value is -1 which means infinity or until the next
subtitle event comes.
milliseconds. Default value is 30000 which is 30 seconds.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.

View File

@@ -25,6 +25,17 @@ Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section apng
Animated Portable Network Graphics demuxer.
@@ -233,16 +244,6 @@ file subdir/file-2.wav
@end example
@end itemize
@section dash
Dynamic Adaptive Streaming over HTTP demuxer.
This demuxer presents all AVStreams found in the manifest.
By setting the discard flags on AVStreams the caller can decide
which streams to actually receive.
Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
@section flv, live_flv
Adobe Flash Video Format demuxer.
@@ -258,12 +259,6 @@ ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@item -flv_ignore_prevtag @var{bool}
Ignore the size of previous tag value.
@item -flv_full_metadata @var{bool}
Output all context of the onMetadata.
@end table
@section gif
@@ -309,15 +304,6 @@ infinitely.
HLS demuxer
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
It accepts the following options:
@table @option
@@ -330,22 +316,6 @@ segment index to start live streams at (negative values are from the end).
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item m3u8_hold_counters
The maximum number of times to load m3u8 when it refreshes without new segments.
Default value is 1000.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams.
Enabled by default.
@item http_multiple
Use multiple HTTP connections for downloading HTTP segments.
Enabled by default for HTTP/1.1 servers.
@item http_seekable
Use HTTP partial requests for downloading HTTP segments.
0 = disable, 1 = enable, -1 = auto, Default is auto.
@end table
@section image2
@@ -456,17 +426,6 @@ nanosecond precision.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@item export_path_metadata
If set to 1, will add two extra fields to the metadata found in input, making them
also available for other filters (see @var{drawtext} filter for examples). Default
value is 0. The extra fields are described below:
@table @option
@item lavf.image2dec.source_path
Corresponds to the full path to the input file being read.
@item lavf.image2dec.source_basename
Corresponds to the name of the file being read.
@end table
@end table
@subsection Examples
@@ -498,84 +457,14 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
The Game Music Emu library is a collection of video game music file emulators.
See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
See @url{http://code.google.com/p/game-music-emu/} for more information.
It accepts the following options:
Some files have multiple tracks. The demuxer will pick the first track by
default. The @option{track_index} option can be used to select a different
track. Track indexes start at 0. The demuxer exports the number of tracks as
@var{tracks} meta data entry.
@table @option
@item track_index
Set the index of which track to demux. The demuxer can only export one track.
Track indexes start at 0. Default is to pick the first track. Number of tracks
is exported as @var{tracks} metadata entry.
@item sample_rate
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
@item max_size @emph{(bytes)}
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read.
Default is 50 MiB.
@end table
@section libmodplug
ModPlug based module demuxer
See @url{https://github.com/Konstanty/libmodplug}
It will export one 2-channel 16-bit 44.1 kHz audio stream.
Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
It accepts the following options:
@table @option
@item noise_reduction
Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
@item reverb_depth
Set amount of reverb. Range 0-100. Default is 0.
@item reverb_delay
Set delay in ms, clamped to 40-250 ms. Default is 0.
@item bass_amount
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
@item bass_range
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
@item surround_depth
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
@item surround_delay
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
@item max_size
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
0 removes buffer size limit (not recommended). Default is 5 MiB.
@item video_stream_expr
String which is evaluated using the eval API to assign colors to the generated video stream.
Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
@code{tempo}, @code{order}, @code{pattern} and @code{row}.
@item video_stream
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
@item video_stream_w
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_h
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_ptxt
Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
@end table
For very large files, the @option{max_size} option may have to be adjusted.
@section libopenmpt
@@ -604,13 +493,9 @@ Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp
@section mov/mp4/3gp/QuickTime
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
@subsection Options
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
@table @option
@@ -621,73 +506,10 @@ Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default.
Enabling this poses a security risk. It should only be enabled if the source
is known to be non-malicious.
is known to be non malicious.
@item seek_streams_individually
When seeking, identify the closest point in each stream individually and demux packets in
that stream from identified point. This can lead to a different sequence of packets compared
to demuxing linearly from the beginning. Default is true.
@item ignore_editlist
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
timeline described by the edit list. Default is false.
@item advanced_editlist
Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
must be set to false for this option to be effective.
If both @code{ignore_editlist} and this option are set to false, then only the
start of the stream index is modified to reflect initial dwell time or starting timestamp
described by the edit list. Default is true.
@item ignore_chapters
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
only parsed when input is seekable. Default is false.
@item use_mfra_for
For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
Following options are available:
@table @samp
@item auto
Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
@item dts
Set mfra timestamps as DTS
@item pts
Set mfra timestamps as PTS
@item 0
Don't use mfra box to set timestamps
@end table
@item export_all
Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
characters of the box type are set as the key. Default is false.
@item export_xmp
Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
but with key @code{XMP_}. Default is false.
@item activation_bytes
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
@item audible_fixed_key
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
specify.
@item decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@subsection Audible AAX
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
@example
ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
@end example
@section mpegts
MPEG-2 transport stream demuxer.
@@ -698,9 +520,6 @@ This demuxer accepts the following options:
Set size limit for looking up a new synchronization. Default value is
65536.
@item skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.
@item fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
@@ -715,10 +534,6 @@ Show the detected raw packet size, cannot be set by the user.
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@item merge_pmt_versions
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@end table
@section mpjpeg
@@ -816,20 +631,4 @@ Example: convert the captions to a format most players understand:
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@section vapoursynth
Vapoursynth wrapper.
Due to security concerns, Vapoursynth scripts will not
be autodetected so the input format has to be forced. For ff* CLI tools,
add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
This demuxer accepts the following option:
@table @option
@item max_script_size
The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@c man end DEMUXERS

View File

@@ -1,79 +0,0 @@
# FFmpeg project
## Organisation
The FFmpeg project is organized through a community working on global consensus.
Decisions are taken by the ensemble of active members, through voting and
are aided by two committees.
## General Assembly
The ensemble of active members is called the General Assembly (GA).
The General Assembly is sovereign and legitimate for all its decisions
regarding the FFmpeg project.
The General Assembly is made up of active contributors.
Contributors are considered "active contributors" if they have pushed more
than 20 patches in the last 36 months in the main FFmpeg repository, or
if they have been voted in by the GA.
Additional members are added to the General Assembly through a vote after
proposal by a member of the General Assembly.
They are part of the GA for two years, after which they need a confirmation by
the GA.
## Voting
Voting is done using a ranked voting system, currently running on https://vote.ffmpeg.org/ .
Majority vote means more than 50% of the expressed ballots.
## Technical Committee
The Technical Committee (TC) is here to arbitrate and make decisions when
technical conflicts occur in the project.
They will consider the merits of all the positions, judge them and make a
decision.
The TC resolves technical conflicts but is not a technical steering committee.
Decisions by the TC are binding for all the contributors.
Decisions made by the TC can be re-opened after 1 year or by a majority vote
of the General Assembly, requested by one of the member of the GA.
The TC is elected by the General Assembly for a duration of 1 year, and
is composed of 5 members.
Members can be re-elected if they wish. A majority vote in the General Assembly
can trigger a new election of the TC.
The members of the TC can be elected from outside of the GA.
Candidates for election can either be suggested or self-nominated.
The conflict resolution process is detailed in the [resolution process](resolution_process.md) document.
## Community committee
The Community Committee (CC) is here to arbitrage and make decisions when
inter-personal conflicts occur in the project. It will decide quickly and
take actions, for the sake of the project.
The CC can remove privileges of offending members, including removal of
commit access and temporary ban from the community.
Decisions made by the CC can be re-opened after 1 year or by a majority vote
of the General Assembly. Indefinite bans from the community must be confirmed
by the General Assembly, in a majority vote.
The CC is elected by the General Assembly for a duration of 1 year, and is
composed of 5 members.
Members can be re-elected if they wish. A majority vote in the General Assembly
can trigger a new election of the CC.
The members of the CC can be elected from outside of the GA.
Candidates for election can either be suggested or self-nominated.
The CC is governed by and responsible for enforcing the Code of Conduct.

View File

@@ -1,91 +0,0 @@
# Technical Committee
_This document only makes sense with the rules from [the community document](community)_.
The Technical Committee (**TC**) is here to arbitrate and make decisions when
technical conflicts occur in the project.
The TC main role is to resolve technical conflicts.
It is therefore not a technical steering committee, but it is understood that
some decisions might impact the future of the project.
# Process
## Seizing
The TC can take possession of any technical matter that it sees fit.
To involve the TC in a matter, email tc@ or CC them on an ongoing discussion.
As members of TC are developers, they also can email tc@ to raise an issue.
## Announcement
The TC, once seized, must announce itself on the main mailing list, with a _[TC]_ tag.
The TC has 2 modes of operation: a RFC one and an internal one.
If the TC thinks it needs the input from the larger community, the TC can call
for a RFC. Else, it can decide by itself.
If the disagreement involves a member of the TC, that member should recuse
themselves from the decision.
The decision to use a RFC process or an internal discussion is a discretionary
decision of the TC.
The TC can also reject a seizure for a few reasons such as:
the matter was not discussed enough previously; it lacks expertise to reach a
beneficial decision on the matter; or the matter is too trivial.
### RFC call
In the RFC mode, one person from the TC posts on the mailing list the
technical question and will request input from the community.
The mail will have the following specification:
* a precise title
* a specific tag [TC RFC]
* a top-level email
* contain a precise question that does not exceed 100 words and that is answerable by developers
* may have an extra description, or a link to a previous discussion, if deemed necessary,
* contain a precise end date for the answers.
The answers from the community must be on the main mailing list and must have
the following specification:
* keep the tag and the title unchanged
* limited to 400 words
* a first-level, answering directly to the main email
* answering to the question.
Further replies to answers are permitted, as long as they conform to the
community standards of politeness, they are limited to 100 words, and are not
nested more than once. (max-depth=2)
After the end-date, mails on the thread will be ignored.
Violations of those rules will be escalated through the Community Committee.
After all the emails are in, the TC has 96 hours to give its final decision.
Exceptionally, the TC can request an extra delay, that will be notified on the
mailing list.
### Within TC
In the internal case, the TC has 96 hours to give its final decision.
Exceptionally, the TC can request an extra delay.
## Decisions
The decisions from the TC will be sent on the mailing list, with the _[TC]_ tag.
Internally, the TC should take decisions with a majority, or using
ranked-choice voting.
The decision from the TC should be published with a summary of the reasons that
lead to this decision.
The decisions from the TC are final, until the matters are reopened after
no less than one year.

View File

@@ -10,7 +10,9 @@
@contents
@chapter Notes for external developers
@chapter Developers Guide
@section Notes for external developers
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
@@ -28,13 +30,15 @@ For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@chapter Contributing
@section Contributing
There are 2 ways by which code gets into FFmpeg:
There are 3 ways by which code gets into FFmpeg.
@itemize @bullet
@item Submitting patches to the ffmpeg-devel mailing list.
@item Submitting patches to the main developer mailing list.
See @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
@@ -43,9 +47,9 @@ The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section Coding Rules
@section Code formatting conventions
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@@ -70,7 +74,7 @@ The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
@@ -110,7 +114,7 @@ int myfunc(int my_parameter)
...
@end example
@section C language features
@subsection C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@@ -128,12 +132,6 @@ designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
@item
for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@item
Implementation defined behavior for signed integers is assumed to match the
expected behavior for two's complement. Non representable values in integer
@@ -162,7 +160,7 @@ mixing statements and declarations;
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
@subsection Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
@@ -186,7 +184,7 @@ e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@samp{avpriv_aac_parse_header}.
@item
Each library has its own prefix for public symbols, in addition to the
@@ -206,7 +204,7 @@ letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@subsection Miscellaneous conventions
@itemize @bullet
@item
@@ -218,7 +216,7 @@ Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@section Editor configuration
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
@@ -251,9 +249,9 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Development Policy
@section Patches/Committing
@subsection Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
@@ -352,7 +350,7 @@ time-frame (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@section Code
@subsection Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
@@ -383,29 +381,12 @@ Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
It is important to be subscribed to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
want to subscribe and discuss the patch, then please do send it to the list
anyway.
@subsection Documentation/Other
@subheading Subscribe to the ffmpeg-cvslog mailing list.
Diffs of all commits are sent to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
It is important to do this as the diffs of all commits are sent there and
reviewed by all the other developers. Bugs and possible improvements or
general questions regarding commits are discussed there. We expect you to
react if problems with your code are uncovered.
@subheading Keep the documentation up to date.
Update the documentation if you change behavior or add features. If you are
@@ -425,7 +406,7 @@ finding a new maintainer and also don't forget to update the @file{MAINTAINERS}
We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
@section Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
@@ -455,7 +436,7 @@ Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@chapter Submitting patches
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
@@ -504,7 +485,7 @@ Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@chapter New codecs or formats checklist
@section New codecs or formats checklist
@enumerate
@item
@@ -556,7 +537,7 @@ Did you make sure it compiles standalone, i.e. with
@end enumerate
@chapter Patch submission checklist
@section patch submission checklist
@enumerate
@item
@@ -566,9 +547,9 @@ Does @code{make fate} pass with the patch applied?
Was the patch generated with git format-patch or send-email?
@item
Did you sign-off your patch? (@code{git commit -s})
See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
of @dfn{sign-off}.
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
@@ -625,7 +606,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug?
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to @url{https://streams.videolan.org/upload/}.
URL, you can upload to ftp://upload.ffmpeg.org.
@item
Did you provide a verbose summary about what the patch does change?
@@ -669,7 +650,7 @@ Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@chapter Patch review process
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
@@ -700,7 +681,7 @@ to be reviewed, please consider helping to review other patches, that is a great
way to get everyone's patches reviewed sooner.
@anchor{Regression tests}
@chapter Regression tests
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
@@ -711,7 +692,7 @@ Running 'make fate' accomplishes this, please see @url{fate.html} for details.
this case, the reference results of the regression tests shall be modified
accordingly].
@section Adding files to the fate-suite dataset
@subsection Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
@@ -722,7 +703,7 @@ Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@section Visualizing Test Coverage
@subsection Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
@@ -749,7 +730,7 @@ You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@section Using Valgrind
@subsection Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@@ -763,7 +744,7 @@ In case you need finer control over how valgrind is invoked, use the
your configure line instead.
@anchor{Release process}
@chapter Release process
@section Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
@@ -795,7 +776,7 @@ adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@section Criteria for Point Releases
@subsection Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@@ -819,7 +800,7 @@ point releases of the same release branch.
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@section Release Checklist
@subsection Release Checklist
The release process involves the following steps:

File diff suppressed because it is too large Load Diff

View File

@@ -1,4 +1,4 @@
/avio_list_dir
/avio_dir_cmd
/avio_reading
/decode_audio
/decode_video
@@ -20,5 +20,3 @@
/scaling_video
/transcode_aac
/transcoding
/vaapi_encode
/vaapi_transcode

View File

@@ -1,4 +1,4 @@
EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
@@ -19,8 +19,6 @@ EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
@@ -37,7 +35,7 @@ $(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
examples: $(EXAMPLES)
$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
OUTDIRS += doc/examples
OBJDIRS += doc/examples
DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)

View File

@@ -11,7 +11,7 @@ CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_list_dir \
EXAMPLES= avio_dir_cmd \
avio_reading \
decode_audio \
decode_video \

View File

@@ -102,15 +102,38 @@ static int list_op(const char *input_dir)
return ret;
}
static int del_op(const char *url)
{
int ret = avpriv_io_delete(url);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
return ret;
}
static int move_op(const char *src, const char *dst)
{
int ret = avpriv_io_move(src, dst);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
return ret;
}
static void usage(const char *program_name)
{
fprintf(stderr, "usage: %s input_dir\n"
"API example program to show how to list files in directory "
"accessed through AVIOContext.\n", program_name);
fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
"API example program to show how to manipulate resources "
"accessed through AVIOContext.\n"
"OPERATIONS:\n"
"list list content of the directory\n"
"move rename content in directory\n"
"del delete content in directory\n",
program_name);
}
int main(int argc, char *argv[])
{
const char *op = NULL;
int ret;
av_log_set_level(AV_LOG_DEBUG);
@@ -120,9 +143,36 @@ int main(int argc, char *argv[])
return 1;
}
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
avformat_network_init();
ret = list_op(argv[1]);
op = argv[1];
if (strcmp(op, "list") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = list_op(argv[2]);
}
} else if (strcmp(op, "del") == 0) {
if (argc < 3) {
av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = del_op(argv[2]);
}
} else if (strcmp(op, "move") == 0) {
if (argc < 4) {
av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
ret = AVERROR(EINVAL);
} else {
ret = move_op(argv[2], argv[3]);
}
} else {
av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
ret = AVERROR(EINVAL);
}
avformat_network_deinit();

View File

@@ -44,8 +44,6 @@ static int read_packet(void *opaque, uint8_t *buf, int buf_size)
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
@@ -74,6 +72,9 @@ int main(int argc, char *argv[])
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
@@ -117,12 +118,11 @@ int main(int argc, char *argv[])
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx)
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
avio_context_free(&avio_ctx);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {

View File

@@ -39,35 +39,6 @@
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
FILE *outfile)
{
@@ -115,9 +86,6 @@ int main(int argc, char **argv)
size_t data_size;
AVPacket *pkt;
AVFrame *decoded_frame = NULL;
enum AVSampleFormat sfmt;
int n_channels = 0;
const char *fmt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
@@ -126,6 +94,9 @@ int main(int argc, char **argv)
filename = argv[1];
outfilename = argv[2];
/* register all the codecs */
avcodec_register_all();
pkt = av_packet_alloc();
/* find the MPEG audio decoder */
@@ -204,26 +175,6 @@ int main(int argc, char **argv)
pkt->size = 0;
decode(c, pkt, decoded_frame, outfile);
/* print output pcm infomations, because there have no metadata of pcm */
sfmt = c->sample_fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
}
n_channels = c->channels;
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, c->sample_rate,
outfilename);
end:
fclose(outfile);
fclose(f);

View File

@@ -41,7 +41,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
FILE *f;
int i;
f = fopen(filename,"wb");
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
@@ -95,13 +95,14 @@ int main(int argc, char **argv)
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n"
"And check your input file is encoded by mpeg1video please.\n", argv[0]);
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
avcodec_register_all();
pkt = av_packet_alloc();
if (!pkt)
exit(1);

View File

@@ -51,97 +51,99 @@ static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket *pkt = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
/* Enable or disable frame reference counting. You are not supposed to support
* both paths in your application but pick the one most appropriate to your
* needs. Look for the use of refcount in this example to see what are the
* differences of API usage between them. */
static int refcount = 0;
printf("video_frame n:%d coded_n:%d\n",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
return 0;
}
static int output_audio_frame(AVFrame *frame)
{
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame n:%d nb_samples:%d pts:%s\n",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
return 0;
}
static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
*got_frame = 0;
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
// those two return values are special and mean there is no output
// frame available, but there were no errors during decoding
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
return 0;
fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
if (*got_frame) {
av_frame_unref(frame);
if (ret < 0)
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame%s n:%d coded_n:%d\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
return 0;
/* If we use frame reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && refcount)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
@@ -184,7 +186,8 @@ static int open_codec_context(int *stream_idx,
return ret;
}
/* Init the decoders */
/* Init the decoders, with or without reference counting */
av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
@@ -227,21 +230,31 @@ static int get_format_from_sample_fmt(const char **fmt,
int main (int argc, char **argv)
{
int ret = 0;
int ret = 0, got_frame;
if (argc != 4) {
fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n",
argv[0]);
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5 && !strcmp(argv[1], "-refcount")) {
refcount = 1;
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
@@ -303,12 +316,10 @@ int main (int argc, char **argv)
goto end;
}
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate packet\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
@@ -316,23 +327,24 @@ int main (int argc, char **argv)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt->stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, pkt);
else if (pkt->stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, pkt);
av_packet_unref(pkt);
if (ret < 0)
break;
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_packet_unref(&orig_pkt);
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
@@ -374,7 +386,6 @@ end:
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_packet_free(&pkt);
av_frame_free(&frame);
av_free(video_dst_data[0]);

View File

@@ -138,6 +138,9 @@ int main(int argc, char **argv)
}
filename = argv[1];
/* register all the codecs */
avcodec_register_all();
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {

View File

@@ -84,6 +84,8 @@ int main(int argc, char **argv)
filename = argv[1];
codec_name = argv[2];
avcodec_register_all();
/* find the mpeg1video encoder */
codec = avcodec_find_encoder_by_name(codec_name);
if (!codec) {
@@ -145,7 +147,7 @@ int main(int argc, char **argv)
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 0);
ret = av_frame_get_buffer(frame, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
@@ -186,8 +188,7 @@ int main(int argc, char **argv)
encode(c, NULL, pkt, f);
/* add sequence end code to have a real MPEG file */
if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
fwrite(endcode, 1, sizeof(endcode), f);
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_free_context(&c);

View File

@@ -129,6 +129,8 @@ int main(int argc, char **argv)
}
src_filename = argv[1];
av_register_all();
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);

View File

@@ -64,13 +64,13 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
const AVFilter *abuffer;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
const AVFilter *volume;
AVFilter *volume;
AVFilterContext *aformat_ctx;
const AVFilter *aformat;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
const AVFilter *abuffersink;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
@@ -289,6 +289,8 @@ int main(int argc, char *argv[])
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {

View File

@@ -32,6 +32,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -74,6 +75,7 @@ static int open_input_file(const char *filename)
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -88,8 +90,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
@@ -227,6 +229,9 @@ int main(int argc, char **argv)
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)

View File

@@ -29,11 +29,10 @@
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -79,6 +78,7 @@ static int open_input_file(const char *filename)
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -93,8 +93,8 @@ static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
@@ -211,20 +211,20 @@ int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
@@ -252,25 +252,27 @@ int main(int argc, char **argv)
goto end;
}
frame->pts = frame->best_effort_timestamp;
if (ret >= 0) {
frame->pts = frame->best_effort_timestamp;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(&packet);

View File

@@ -114,6 +114,7 @@ int main(int argc, char **argv)
in_uri = argv[1];
out_uri = argv[2];
av_register_all();
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {

View File

@@ -4,23 +4,21 @@
*
* HW Acceleration API (video decoding) decode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
* This file is part of FFmpeg.
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
@@ -46,6 +44,34 @@ static AVBufferRef *hw_device_ctx = NULL;
static enum AVPixelFormat hw_pix_fmt;
static FILE *output_file = NULL;
static enum AVPixelFormat find_fmt_by_hw_type(const enum AVHWDeviceType type)
{
enum AVPixelFormat fmt;
switch (type) {
case AV_HWDEVICE_TYPE_VAAPI:
fmt = AV_PIX_FMT_VAAPI;
break;
case AV_HWDEVICE_TYPE_DXVA2:
fmt = AV_PIX_FMT_DXVA2_VLD;
break;
case AV_HWDEVICE_TYPE_D3D11VA:
fmt = AV_PIX_FMT_D3D11;
break;
case AV_HWDEVICE_TYPE_VDPAU:
fmt = AV_PIX_FMT_VDPAU;
break;
case AV_HWDEVICE_TYPE_VIDEOTOOLBOX:
fmt = AV_PIX_FMT_VIDEOTOOLBOX;
break;
default:
fmt = AV_PIX_FMT_NONE;
break;
}
return fmt;
}
static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
{
int err = 0;
@@ -88,7 +114,7 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
return ret;
}
while (1) {
while (ret >= 0) {
if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
fprintf(stderr, "Can not alloc frame\n");
ret = AVERROR(ENOMEM);
@@ -140,10 +166,13 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
fail:
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_freep(&buffer);
if (buffer)
av_freep(&buffer);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char *argv[])
@@ -155,20 +184,18 @@ int main(int argc, char *argv[])
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
if (argc < 4) {
fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
fprintf(stderr, "Usage: %s <vaapi|vdpau|dxva2|d3d11va> <input file> <output file>\n", argv[0]);
return -1;
}
av_register_all();
type = av_hwdevice_find_type_by_name(argv[1]);
if (type == AV_HWDEVICE_TYPE_NONE) {
fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
fprintf(stderr, "Available device types:");
while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
fprintf(stderr, "\n");
hw_pix_fmt = find_fmt_by_hw_type(type);
if (hw_pix_fmt == -1) {
fprintf(stderr, "Cannot support '%s' in this example.\n", argv[1]);
return -1;
}
@@ -191,20 +218,6 @@ int main(int argc, char *argv[])
}
video_stream = ret;
for (i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
if (!config) {
fprintf(stderr, "Decoder %s does not support device type %s.\n",
decoder->name, av_hwdevice_get_type_name(type));
return -1;
}
if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
config->device_type == type) {
hw_pix_fmt = config->pix_fmt;
break;
}
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
@@ -213,6 +226,7 @@ int main(int argc, char *argv[])
return -1;
decoder_ctx->get_format = get_hw_format;
av_opt_set_int(decoder_ctx, "refcounted_frames", 1, 0);
if (hw_decoder_init(decoder_ctx, type) < 0)
return -1;
@@ -223,7 +237,7 @@ int main(int argc, char *argv[])
}
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+b");
output_file = fopen(argv[3], "w+");
/* actual decoding and dump the raw data */
while (ret >= 0) {

View File

@@ -44,14 +44,10 @@ int main (int argc, char **argv)
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);

View File

@@ -78,45 +78,15 @@ static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame)
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
int ret;
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
// send the frame to the encoder
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
AVPacket pkt = { 0 };
ret = avcodec_receive_packet(c, &pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
exit(1);
}
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, &pkt);
ret = av_interleaved_write_frame(fmt_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
}
}
return ret == AVERROR_EOF ? 1 : 0;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
@@ -200,7 +170,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
break;
default:
break;
@@ -284,25 +254,25 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -315,7 +285,7 @@ static AVFrame *get_audio_frame(OutputStream *ost)
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
@@ -339,20 +309,23 @@ static AVFrame *get_audio_frame(OutputStream *ost)
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
@@ -376,7 +349,22 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
@@ -396,7 +384,7 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 0);
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
@@ -476,7 +464,7 @@ static AVFrame *get_video_frame(OutputStream *ost)
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
@@ -500,9 +488,9 @@ static AVFrame *get_video_frame(OutputStream *ost)
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
@@ -518,7 +506,37 @@ static AVFrame *get_video_frame(OutputStream *ost)
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost));
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
frame = get_video_frame(ost);
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
@@ -546,6 +564,9 @@ int main(int argc, char **argv)
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"

View File

@@ -150,6 +150,8 @@ int main(int argc, char **argv)
int ret, i;
av_register_all();
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
@@ -208,6 +210,7 @@ int main(int argc, char **argv)
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->refcounted_frames = 1;
decoder_ctx->opaque = &decode;
decoder_ctx->get_format = get_format;

View File

@@ -65,6 +65,8 @@ int main(int argc, char **argv)
in_filename = argv[1];
out_filename = argv[2];
av_register_all();
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;

View File

@@ -1,6 +1,4 @@
/*
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -10,7 +8,7 @@
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
@@ -20,11 +18,10 @@
/**
* @file
* Simple audio converter
* simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
@@ -43,18 +40,12 @@
#include "libswresample/swresample.h"
/* The output bit rate in bit/s */
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 96000
/* The number of output channels */
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context Codec context of opened file
* @return Error code (0 if successful)
*/
/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
@@ -63,7 +54,7 @@ static int open_input_file(const char *filename,
AVCodec *input_codec;
int error;
/* Open the input file to read from it. */
/** Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
@@ -72,7 +63,7 @@ static int open_input_file(const char *filename,
return error;
}
/* Get information on the input file (number of streams etc.). */
/** Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
av_err2str(error));
@@ -80,7 +71,7 @@ static int open_input_file(const char *filename,
return error;
}
/* Make sure that there is only one stream in the input file. */
/** Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
@@ -88,14 +79,14 @@ static int open_input_file(const char *filename,
return AVERROR_EXIT;
}
/* Find a decoder for the audio stream. */
/** Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/* Allocate a new decoding context. */
/** allocate a new decoding context */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
@@ -103,7 +94,7 @@ static int open_input_file(const char *filename,
return AVERROR(ENOMEM);
}
/* Initialize the stream parameters with demuxer information. */
/** initialize the stream parameters with demuxer information */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
@@ -111,7 +102,7 @@ static int open_input_file(const char *filename,
return error;
}
/* Open the decoder for the audio stream to use it later. */
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
av_err2str(error));
@@ -120,7 +111,7 @@ static int open_input_file(const char *filename,
return error;
}
/* Save the decoder context for easier access later. */
/** Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
@@ -130,11 +121,6 @@ static int open_input_file(const char *filename,
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param filename File to be opened
* @param input_codec_context Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
@@ -147,7 +133,7 @@ static int open_output_file(const char *filename,
AVCodec *output_codec = NULL;
int error;
/* Open the output file to write to it. */
/** Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
@@ -155,35 +141,32 @@ static int open_output_file(const char *filename,
return error;
}
/* Create a new format context for the output container format. */
/** Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/* Associate the output file (pointer) with the container format context. */
/** Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/* Guess the desired container format based on the file extension. */
/** Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
if (!((*output_format_context)->url = av_strdup(filename))) {
fprintf(stderr, "Could not allocate url.\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/* Find the encoder to be used by its name. */
/** Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/* Create a new audio stream in the output file container. */
/** Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
@@ -197,27 +180,31 @@ static int open_output_file(const char *filename,
goto cleanup;
}
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/* Allow the use of the experimental AAC encoder. */
/** Allow the use of the experimental AAC encoder */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
/** Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/* Some container formats (like MP4) require global headers to be present.
* Mark the encoder so that it behaves accordingly. */
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Open the encoder for the audio stream to use it later. */
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
@@ -230,7 +217,7 @@ static int open_output_file(const char *filename,
goto cleanup;
}
/* Save the encoder context for easier access later. */
/** Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
@@ -243,25 +230,16 @@ cleanup:
return error < 0 ? error : AVERROR_EXIT;
}
/**
* Initialize one data packet for reading or writing.
* @param[out] packet Packet to be initialized
* @return Error code (0 if successful)
*/
static int init_packet(AVPacket **packet)
/** Initialize one data packet for reading or writing. */
static void init_packet(AVPacket *packet)
{
if (!(*packet = av_packet_alloc())) {
fprintf(stderr, "Could not allocate packet\n");
return AVERROR(ENOMEM);
}
return 0;
av_init_packet(packet);
/** Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/**
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
@@ -275,10 +253,6 @@ static int init_input_frame(AVFrame **frame)
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
@@ -286,7 +260,7 @@ static int init_resampler(AVCodecContext *input_codec_context,
{
int error;
/*
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
@@ -305,14 +279,14 @@ static int init_resampler(AVCodecContext *input_codec_context,
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/*
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the resampler with the specified parameters. */
/** Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
@@ -321,15 +295,10 @@ static int init_resampler(AVCodecContext *input_codec_context,
return 0;
}
/**
* Initialize a FIFO buffer for the audio samples to be encoded.
* @param[out] fifo Sample buffer
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
@@ -338,11 +307,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
return 0;
}
/**
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
/** Write the header of the output file container. */
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
@@ -354,90 +319,57 @@ static int write_output_file_header(AVFormatContext *output_format_context)
return 0;
}
/**
* Decode one audio frame from the input file.
* @param frame Audio frame to be decoded
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param[out] data_present Indicates whether data has been decoded
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false, there
* is more data to be decoded, i.e., this
* function has to be called again.
* @return Error code (0 if successful)
*/
/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/* Packet used for temporary storage. */
AVPacket *input_packet;
/** Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
error = init_packet(&input_packet);
if (error < 0)
return error;
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
av_err2str(error));
goto cleanup;
return error;
}
}
/* Send the audio frame stored in the temporary packet to the decoder.
* The input audio stream decoder is used to do this. */
if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
av_err2str(error));
goto cleanup;
}
/* Receive one frame from the decoder. */
error = avcodec_receive_frame(input_codec_context, frame);
/* If the decoder asks for more data to be able to decode a frame,
* return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the end of the input file is reached, stop decoding. */
} else if (error == AVERROR_EOF) {
*finished = 1;
error = 0;
goto cleanup;
} else if (error < 0) {
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return decoded data. */
} else {
*data_present = 1;
goto cleanup;
av_packet_unref(&input_packet);
return error;
}
cleanup:
av_packet_free(&input_packet);
return error;
/**
* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again.
*/
if (*finished && *data_present)
*finished = 0;
av_packet_unref(&input_packet);
return 0;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
* dimensions are reference, channel
* (for multi-channel audio), sample.
* @param output_codec_context Codec context of the output file
* @param frame_size Number of samples to be converted in
* each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
@@ -445,7 +377,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
{
int error;
/* Allocate as many pointers as there are audio channels.
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
@@ -455,8 +388,10 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
@@ -473,15 +408,8 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param input_data Samples to be decoded. The dimensions are
* channel (for multi-channel audio), sample.
* @param[out] converted_data Converted samples. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
@@ -489,7 +417,7 @@ static int convert_samples(const uint8_t **input_data,
{
int error;
/* Convert the samples using the resampler. */
/** Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
@@ -501,28 +429,23 @@ static int convert_samples(const uint8_t **input_data,
return 0;
}
/**
* Add converted input audio samples to the FIFO buffer for later processing.
* @param fifo Buffer to add the samples to
* @param converted_input_samples Samples to be added. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @return Error code (0 if successful)
*/
/** Add converted input audio samples to the FIFO buffer for later processing. */
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples. */
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/* Store the new samples in the FIFO buffer. */
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
@@ -532,20 +455,8 @@ static int add_samples_to_fifo(AVAudioFifo *fifo,
}
/**
* Read one audio frame from the input file, decode, convert and store
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
* @param fifo Buffer used for temporary storage
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param resampler_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
* there is more data to be decoded,
* i.e., this function has to be called
* again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
@@ -554,41 +465,45 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo,
SwrContext *resampler_context,
int *finished)
{
/* Temporary storage of the input samples of the frame read from the file. */
/** Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
/** Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present = 0;
int data_present;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
/** Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/* Decode one frame worth of audio samples. */
/** Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/* If we are at the end of the file and there are no more samples
/**
* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error. */
if (*finished) {
* This must not be treated as an error.
*/
if (*finished && !data_present) {
ret = 0;
goto cleanup;
}
/* If there is decoded data, convert and store it. */
/** If there is decoded data, convert and store it */
if (data_present) {
/* Initialize the temporary storage for the converted input samples. */
/** Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/* Add the converted input samples to the FIFO buffer for later processing. */
/** Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
@@ -609,10 +524,6 @@ cleanup:
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame Frame to be initialized
* @param output_codec_context Codec context of the output file
* @param frame_size Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
@@ -620,24 +531,28 @@ static int init_output_frame(AVFrame **frame,
{
int error;
/* Create a new frame to store the audio samples. */
/** Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/* Set the frame's parameters, especially its size and format.
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
* are assumed for simplicity.
*/
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified. */
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
@@ -648,117 +563,87 @@ static int init_output_frame(AVFrame **frame,
return 0;
}
/* Global timestamp for the audio frames. */
/** Global timestamp for the audio frames */
static int64_t pts = 0;
/**
* Encode one frame worth of audio to the output file.
* @param frame Samples to be encoded
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @param[out] data_present Indicates whether data has been
* encoded
* @return Error code (0 if successful)
*/
/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/* Packet used for temporary storage. */
AVPacket *output_packet;
/** Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
error = init_packet(&output_packet);
if (error < 0)
return error;
/* Set a timestamp based on the sample rate for the container. */
/** Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
error = avcodec_send_frame(output_codec_context, frame);
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
goto cleanup;
}
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, output_packet);
/* If the encoder asks for more data to be able to provide an
* encoded frame, return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the last frame has been encoded, stop encoding. */
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return encoded data. */
} else {
*data_present = 1;
av_packet_unref(&output_packet);
return error;
}
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present &&
(error = av_write_frame(output_format_context, output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/** Write one audio frame from the temporary packet to the output file. */
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
av_packet_unref(&output_packet);
return error;
}
av_packet_unref(&output_packet);
}
cleanup:
av_packet_free(&output_packet);
return error;
return 0;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/* Temporary storage of the output samples of the frame written to the file. */
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/* Use the maximum number of possible samples per frame.
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size. */
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/* Initialize temporary storage for one output frame. */
/** Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily. */
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/* Encode one frame worth of audio samples. */
/** Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
@@ -768,11 +653,7 @@ static int load_encode_and_write(AVAudioFifo *fifo,
return 0;
}
/**
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
/** Write the trailer of the output file container. */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
@@ -784,6 +665,7 @@ static int write_output_file_trailer(AVFormatContext *output_format_context)
return 0;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
@@ -792,75 +674,90 @@ int main(int argc, char **argv)
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc != 3) {
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/* Open the input file for reading. */
/** Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/* Open the output file for writing. */
/** Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/* Initialize the resampler to be able to convert audio sample formats. */
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/* Initialize the FIFO buffer to store audio samples to be encoded. */
/** Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/* Write the header of the output file container. */
/** Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither. */
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
while (1) {
/* Use the encoder's desired frame size for processing. */
/** Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/* Make sure that there is one frame worth of samples in the FIFO
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples. */
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer. */
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
/**
* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
if (finished)
break;
}
/* If we have enough samples for the encoder, we encode them.
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder. */
* the encoder.
*/
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
if (finished) {
int data_written;
/* Flush the encoder as it may have delayed frames. */
/** Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
@@ -869,7 +766,7 @@ int main(int argc, char **argv)
}
}
/* Write the trailer of the output file container. */
/** Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;

View File

@@ -30,6 +30,7 @@
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
@@ -41,17 +42,12 @@ typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
AVPacket *enc_pkt;
AVFrame *filtered_frame;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
AVFrame *dec_frame;
} StreamContext;
static StreamContext *stream_ctx;
@@ -107,10 +103,6 @@ static int open_input_file(const char *filename)
}
}
stream_ctx[i].dec_ctx = codec_ctx;
stream_ctx[i].dec_frame = av_frame_alloc();
if (!stream_ctx[i].dec_frame)
return AVERROR(ENOMEM);
}
av_dump_format(ifmt_ctx, 0, filename, 0);
@@ -181,9 +173,6 @@ static int open_output_file(const char *filename)
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
@@ -195,6 +184,8 @@ static int open_output_file(const char *filename)
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
@@ -237,8 +228,8 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -407,63 +398,54 @@ static int init_filters(void)
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
filter_ctx[i].enc_pkt = av_packet_alloc();
if (!filter_ctx[i].enc_pkt)
return AVERROR(ENOMEM);
filter_ctx[i].filtered_frame = av_frame_alloc();
if (!filter_ctx[i].filtered_frame)
return AVERROR(ENOMEM);
}
return 0;
}
static int encode_write_frame(unsigned int stream_index, int flush)
{
StreamContext *stream = &stream_ctx[stream_index];
FilteringContext *filter = &filter_ctx[stream_index];
AVFrame *filt_frame = flush ? NULL : filter->filtered_frame;
AVPacket *enc_pkt = filter->enc_pkt;
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
av_packet_unref(enc_pkt);
ret = avcodec_send_frame(stream->enc_ctx, filt_frame);
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
while (ret >= 0) {
ret = avcodec_receive_packet(stream->enc_ctx, enc_pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return 0;
/* prepare packet for muxing */
enc_pkt->stream_index = stream_index;
av_packet_rescale_ts(enc_pkt,
stream->enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
}
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
FilteringContext *filter = &filter_ctx[stream_index];
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter->buffersrc_ctx,
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
@@ -472,9 +454,14 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter->buffersink_ctx,
filter->filtered_frame);
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
@@ -482,12 +469,12 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filter->filtered_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(stream_index, 0);
av_frame_unref(filter->filtered_frame);
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
@@ -497,80 +484,99 @@ static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
return encode_write_frame(stream_index, 1);
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket *packet = NULL;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
if (!(packet = av_packet_alloc()))
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, packet)) < 0)
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet->stream_index;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
StreamContext *stream = &stream_ctx[stream_index];
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
av_packet_rescale_ts(packet,
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
stream->dec_ctx->time_base);
ret = avcodec_send_packet(stream->dec_ctx, packet);
stream_ctx[stream_index].dec_ctx->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(stream->dec_ctx, stream->dec_frame);
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
break;
else if (ret < 0)
goto end;
stream->dec_frame->pts = stream->dec_frame->best_effort_timestamp;
ret = filter_encode_write_frame(stream->dec_frame, stream_index);
if (got_frame) {
frame->pts = frame->best_effort_timestamp;
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(packet,
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, packet);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(packet);
av_packet_unref(&packet);
}
/* flush filters and encoders */
@@ -594,18 +600,14 @@ int main(int argc, char **argv)
av_write_trailer(ofmt_ctx);
end:
av_packet_free(&packet);
av_packet_unref(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
if (filter_ctx && filter_ctx[i].filter_graph) {
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
av_packet_free(&filter_ctx[i].enc_pkt);
av_frame_free(&filter_ctx[i].filtered_frame);
}
av_frame_free(&stream_ctx[i].dec_frame);
}
av_free(filter_ctx);
av_free(stream_ctx);

View File

@@ -1,224 +0,0 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
*/
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
static int width, height;
static AVBufferRef *hw_device_ctx = NULL;
static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
{
AVBufferRef *hw_frames_ref;
AVHWFramesContext *frames_ctx = NULL;
int err = 0;
if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
fprintf(stderr, "Failed to create VAAPI frame context.\n");
return -1;
}
frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
frames_ctx->format = AV_PIX_FMT_VAAPI;
frames_ctx->sw_format = AV_PIX_FMT_NV12;
frames_ctx->width = width;
frames_ctx->height = height;
frames_ctx->initial_pool_size = 20;
if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
fprintf(stderr, "Failed to initialize VAAPI frame context."
"Error code: %s\n",av_err2str(err));
av_buffer_unref(&hw_frames_ref);
return err;
}
ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
if (!ctx->hw_frames_ctx)
err = AVERROR(ENOMEM);
av_buffer_unref(&hw_frames_ref);
return err;
}
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket *enc_pkt;
if (!(enc_pkt = av_packet_alloc()))
return AVERROR(ENOMEM);
if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
fprintf(stderr, "Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(avctx, enc_pkt);
if (ret)
break;
enc_pkt->stream_index = 0;
ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
av_packet_unref(enc_pkt);
}
end:
av_packet_free(&enc_pkt);
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
int main(int argc, char *argv[])
{
int size, err;
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {
fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
return -1;
}
width = atoi(argv[1]);
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
if (!(fout = fopen(argv[4], "w+b"))) {
fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
err = -1;
goto close;
}
err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
NULL, NULL, 0);
if (err < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
goto close;
}
if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
fprintf(stderr, "Could not find encoder.\n");
err = -1;
goto close;
}
if (!(avctx = avcodec_alloc_context3(codec))) {
err = AVERROR(ENOMEM);
goto close;
}
avctx->width = width;
avctx->height = height;
avctx->time_base = (AVRational){1, 25};
avctx->framerate = (AVRational){25, 1};
avctx->sample_aspect_ratio = (AVRational){1, 1};
avctx->pix_fmt = AV_PIX_FMT_VAAPI;
/* set hw_frames_ctx for encoder's AVCodecContext */
if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
fprintf(stderr, "Failed to set hwframe context.\n");
goto close;
}
if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
goto close;
}
while (1) {
if (!(sw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
/* read data into software frame, and transfer them into hw frame */
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;
if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
break;
if (!(hw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
fprintf(stderr, "Error code: %s.\n", av_err2str(err));
goto close;
}
if (!hw_frame->hw_frames_ctx) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
fprintf(stderr, "Error while transferring frame data to surface."
"Error code: %s.\n", av_err2str(err));
goto close;
}
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}
av_frame_free(&hw_frame);
av_frame_free(&sw_frame);
}
/* flush encoder */
err = encode_write(avctx, NULL, fout);
if (err == AVERROR_EOF)
err = 0;
close:
if (fin)
fclose(fin);
if (fout)
fclose(fout);
av_frame_free(&sw_frame);
av_frame_free(&hw_frame);
avcodec_free_context(&avctx);
av_buffer_unref(&hw_device_ctx);
return err;
}

View File

@@ -1,308 +0,0 @@
/*
* Video Acceleration API (video transcoding) transcode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel VAAPI-accelerated transcoding example.
*
* @example vaapi_transcode.c
* This example shows how to do VAAPI-accelerated transcoding.
* Usage: vaapi_transcode input_stream codec output_stream
* e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
* - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
static AVStream *ost;
static int initialized = 0;
static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
if (*p == AV_PIX_FMT_VAAPI)
return *p;
}
fprintf(stderr, "Unable to decode this file using VA-API.\n");
return AV_PIX_FMT_NONE;
}
static int open_input_file(const char *filename)
{
int ret;
AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = ifmt_ctx->streams[video_stream];
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_vaapi_format;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVPacket *enc_pkt, AVFrame *frame)
{
int ret = 0;
av_packet_unref(enc_pkt);
if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(encoder_ctx, enc_pkt);
if (ret)
break;
enc_pkt->stream_index = 0;
av_packet_rescale_ts(enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
ofmt_ctx->streams[0]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, enc_pkt);
if (ret < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return -1;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!initialized) {
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
* xxx: now the sample can't handle resolution change case.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_VAAPI;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
goto fail;
}
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
initialized = 1;
}
if ((ret = encode_write(pkt, frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0;
AVPacket *dec_pkt;
AVCodec *enc_codec;
if (argc != 4) {
fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return -1;
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
return -1;
}
dec_pkt = av_packet_alloc();
if (!dec_pkt) {
fprintf(stderr, "Failed to allocate decode packet\n");
goto end;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, dec_pkt)) < 0)
break;
if (video_stream == dec_pkt->stream_index)
ret = dec_enc(dec_pkt, enc_codec);
av_packet_unref(dec_pkt);
}
/* flush decoder */
av_packet_unref(dec_pkt);
ret = dec_enc(dec_pkt, enc_codec);
/* flush encoder */
ret = encode_write(dec_pkt, NULL);
/* write the trailer for output stream */
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
avformat_close_input(&ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
av_packet_free(&dec_pkt);
return ret;
}

View File

@@ -76,7 +76,7 @@ the gcc developers. Note that we will not add workarounds for gcc bugs.
Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@@ -257,13 +257,13 @@ default.
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
but beware the '-g 100' might cause problems with some decoders.
Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
@@ -501,71 +501,6 @@ ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@anchor{background task}
@section How do I run ffmpeg as a background task?
ffmpeg normally checks the console input, for entries like "q" to stop
and "?" to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
To prevent those input checks, allowing ffmpeg to run as a background task,
use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to @code{/dev/null} (on Linux and macOS)
or @code{NUL} (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
For example:
@example
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, macOS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@end example
or (on Windows):
@example
ffmpeg -i INPUT OUTPUT <NUL
@end example
@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like @emph{suspended (tty output)}. The question is how
to prevent the process from being suspended.
For example:
@example
% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &>
@end example
The message "tty output" notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
or redirect standard input with @code{< /dev/null}.
See FAQ
@ref{background task, @emph{How do I run ffmpeg as a background task?}}
for details.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -601,7 +536,7 @@ No. These tools are too bloated and they complicate the build.
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?

View File

@@ -147,32 +147,6 @@ process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
If you need a sample uploaded send a mail to samples-request.
This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
Also all needed samples for a commit should be uploaded, ideally 24
hours, before the push.
If you need an account for frequently uploading samples or you wish to help
others by doing that send a mail to ffmpeg-devel.
@example
#First update your local samples copy:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
@end example
@chapter FATE makefile targets and variables
@@ -228,11 +202,6 @@ Set to @samp{1} to generate the missing or mismatched references.
Specify which hardware acceleration to use while running regression tests,
by default @samp{none} is used.
@item KEEP
Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
Default is @samp{0}, which removes these files. Files are always kept when a test
fails.
@end table
@section Examples

View File

@@ -26,12 +26,12 @@ bitstream level modifications without performing decoding.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ the libavcodec library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavdevice library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavfilter library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ provided by the libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ libavformat library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi

View File

@@ -25,12 +25,12 @@ and convert audio format and packing layout.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
@end ifnothtml
@include authors.texi

View File

@@ -24,12 +24,12 @@ image rescaling and pixel format conversion.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
@end ifnothtml
@include authors.texi

View File

@@ -23,12 +23,12 @@ by the libavutil library.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
@end ifnothtml
@include authors.texi

View File

@@ -216,208 +216,16 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION
@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
video, audio, subtitle and data streams respectively, whether manually mapped or automatically
selected, except for those streams which are outputs of complex filtergraphs.
By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
present in the input files and adds them to each output file. It picks the
"best" of each based upon the following criteria: for video, it is the stream
with the highest resolution, for audio, it is the stream with the most channels, for
subtitles, it is the first subtitle stream. In the case where several streams of
the same type rate equally, the stream with the lowest index is chosen.
@section Description
The sub-sections that follow describe the various rules that are involved in stream selection.
The examples that follow next show how these rules are applied in practice.
While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
continuous development and the code may have changed since the time of this writing.
@subsection Automatic stream selection
In the absence of any map options for a particular output file, ffmpeg inspects the output
format to check which type of streams can be included in it, viz. video, audio and/or
subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
from among all the inputs.
It will select that stream based upon the following criteria:
@itemize
@item
for video, it is the stream with the highest resolution,
@item
for audio, it is the stream with the most channels,
@item
for subtitles, it is the first subtitle stream found but there's a caveat.
The output format's default subtitle encoder can be either text-based or image-based,
and only a subtitle stream of the same type will be chosen.
@end itemize
In the case where several streams of the same type rate equally, the stream with the lowest
index is chosen.
Data or attachment streams are not automatically selected and can only be included
using @code{-map}.
@subsection Manual stream selection
When @code{-map} is used, only user-mapped streams are included in that output file,
with one possible exception for filtergraph outputs described below.
@subsection Complex filtergraphs
If there are any complex filtergraph output streams with unlabeled pads, they will be added
to the first output file. This will lead to a fatal error if the stream type is not supported
by the output format. In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map options are present,
these filtergraph streams are included in addition to the mapped streams.
Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
@subsection Stream handling
Stream handling is independent of stream selection, with an exception for subtitles described
below. Stream handling is set via the @code{-codec} option addressed to streams within a
specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
stream selection process and thus do not influence the latter. If no @code{-codec} option is
specified for a stream type, ffmpeg will select the default encoder registered by the output
file muxer.
An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
if the specified encoder can convert the selected stream or if the converted stream is acceptable
within the output format. This applies generally as well: when the user sets an encoder manually,
the stream selection process cannot check if the encoded stream can be muxed into the output file.
If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
@section Examples
The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
selection methods.
They assume the following three input files.
@verbatim
input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels
input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)
input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)
@end verbatim
@subsubheading Example: automatic stream selection
@example
ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
@end example
There are three output files specified, and for the first two, no @code{-map} options
are set, so ffmpeg will select streams for these two files automatically.
@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
so ffmpeg will try to select one of each type.@*
For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
resolution among all the input video streams.@*
For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
number of channels.@*
For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
stream from among @file{A.avi} and @file{B.mp4}.
@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
selected.
For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
occur. The @code{-map 1:a} option will select all audio streams from the second input
@file{B.mp4}. No other streams will be included in this output file.
For the first two outputs, all included streams will be transcoded. The encoders chosen will
be the default ones registered by each output format, which may not match the codec of the
selected input streams.
For the third output, codec option for audio streams has been set
to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
Packets of selected streams shall be conveyed from the input file and muxed within the output
file.
@subsubheading Example: automatic subtitles selection
@example
ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
@end example
Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
for the subtitles is expected to fail and hence the stream isn't selected. However, in
@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
selected, in addition to the video stream. The presence of @code{-an} disables audio stream
selection for @file{out2.mkv}.
@subsubheading Example: unlabeled filtergraph outputs
@example
ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
@end example
A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
video filter. The @code{overlay} filter requires exactly two video inputs, but none are
specified, so the first two available video streams are used, those of @file{A.avi} and
@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
@subsubheading Example: labeled filtergraph outputs
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv
@end example
The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
None of the output files shall be processed.
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv
@end example
This command above will also fail as the hue filter output has a label, @code{[outv]},
and hasn't been mapped anywhere.
The command should be modified as follows,
@example
ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv
@end example
The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
output files.
The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
only suppresses automatic or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
automatic stream selection.
@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
stream from @file{B.mp4}.
@*
You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
full manual control, use the @code{-map} option, which disables the defaults just
described.
@c man end STREAM SELECTION
@@ -481,8 +289,8 @@ see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1)
-to and -t are mutually exclusive and -t has priority.
@item -to @var{position} (@emph{input/output})
Stop writing the output or reading the input at @var{position}.
@item -to @var{position} (@emph{output})
Stop writing the output at @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@@ -508,7 +316,7 @@ input until the timestamps reach @var{position}.
@var{position} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -sseof @var{position} (@emph{input})
@item -sseof @var{position} (@emph{input/output})
Like the @code{-ss} option but relative to the "end of file". That is negative
values are earlier in the file, 0 is at EOF.
@@ -523,9 +331,6 @@ The offset is added to the timestamps of the input files. Specifying
a positive offset means that the corresponding streams are delayed by
the time duration specified in @var{offset}.
@item -itsscale @var{scale} (@emph{input,per-stream})
Rescale input timestamps. @var{scale} should be a floating point number.
@item -timestamp @var{date} (@emph{output})
Set the recording timestamp in the container.
@@ -570,31 +375,22 @@ The following dispositions are recognized:
@item hearing_impaired
@item visual_impaired
@item clean_effects
@item attached_pic
@item captions
@item descriptions
@item dependent
@item metadata
@end table
For example, to make the second audio stream the default stream:
@example
ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
ffmpeg -i in.mkv -disposition:a:1 default out.mkv
@end example
To make the second subtitle stream the default stream and remove the default
disposition from the first subtitle stream:
@example
ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
@end example
To add an embedded cover/thumbnail:
@example
ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
@end example
Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -617,111 +413,6 @@ they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
@end example
The parameters set for each target are as follows.
@strong{VCD}
@example
@var{pal}:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x288 -r 25
-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k
@var{ntsc}:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 30000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k
@var{film}:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 24000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k
@end example
@strong{SVCD}
@example
@var{pal}:
-f svcd -packetsize 2324
-s 480x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k
@var{ntsc}:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k
@var{film}:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k
@end example
@strong{DVD}
@example
@var{pal}:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k
@var{ntsc}:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k
@var{film}:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k
@end example
@strong{DV}
@example
@var{pal}:
-f dv
-s 720x576 -pix_fmt yuv420p -r 25
-ar 48000 -ac 2
@var{ntsc}:
-f dv
-s 720x480 -pix_fmt yuv411p -r 30000/1001
-ar 48000 -ac 2
@var{film}:
-f dv
-s 720x480 -pix_fmt yuv411p -r 24000/1001
-ar 48000 -ac 2
@end example
The @code{dv50} target is identical to the @code{dv} target except that the pixel format set is @code{yuv422p} for all three standards.
Any user-set value for a parameter above will override the target preset value. In that case, the output may
not comply with the target standard.
@item -dn (@emph{input/output})
As an input option, blocks all data streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables data recording i.e. automatic selection or
mapping of any data stream. For full manual control see the @code{-map}
option.
@item -dframes @var{number} (@emph{output})
Set the number of data frames to output. This is an obsolete alias for
@code{-frames:d}, which you should use instead.
@@ -771,20 +462,14 @@ Specify the preset for matching stream(s).
Print encoding progress/statistics. It is on by default, to explicitly
disable it you need to specify @code{-nostats}.
@item -stats_period @var{time} (@emph{global})
Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.
@item -progress @var{url} (@emph{global})
Send program-friendly progress information to @var{url}.
Progress information is written periodically and at the end of
Progress information is written approximately every second and at the end of
the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
consists of only alphanumeric characters. The last key of a sequence of
progress information is always "progress".
The update period is set using @code{-stats_period}.
@anchor{stdin option}
@item -stdin
Enable interaction on standard input. On by default unless standard input is
used as an input. To explicitly disable interaction you need to specify
@@ -835,6 +520,10 @@ ffmpeg -dump_attachment:t "" -i INPUT
Technical note -- attachments are implemented as codec extradata, so this
option can actually be used to extract extradata from any stream, not just
attachments.
@item -noautorotate
Disable automatically rotating video based on file metadata.
@end table
@section Video Options
@@ -855,13 +544,6 @@ If in doubt use @option{-framerate} instead of the input option @option{-r}.
As an output option, duplicate or drop input frames to achieve constant output
frame rate @var{fps}.
@item -fpsmax[:@var{stream_specifier}] @var{fps} (@emph{output,per-stream})
Set maximum frame rate (Hz value, fraction or abbreviation).
Clamps output frame rate when output framerate is auto-set and is higher than this value.
Useful in batch processing or when input framerate is wrongly detected as very high.
It cannot be set together with @code{-r}. It is ignored during streamcopy.
@item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
Set frame size.
@@ -887,14 +569,8 @@ If used together with @option{-vcodec copy}, it will affect the aspect ratio
stored at container level, but not the aspect ratio stored in encoded
frames, if it exists.
@item -vn (@emph{input/output})
As an input option, blocks all video streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables video recording i.e. automatic selection or
mapping of any video stream. For full manual control see the @code{-map}
option.
@item -vn (@emph{output})
Disable video recording.
@item -vcodec @var{codec} (@emph{output})
Set the video codec. This is an alias for @code{-codec:v}.
@@ -923,18 +599,6 @@ Create the filtergraph specified by @var{filtergraph} and use it to
filter the stream.
This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@item -autorotate
Automatically rotate the video according to file metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -autoscale
Automatically scale the video according to the resolution of first frame.
Enabled by default, use @option{-noautoscale} to disable it. When autoscale is
disabled, all output frames of filter graph might not be in the same resolution
and may be inadequate for some encoder/muxer. Therefore, it is not recommended
to disable it unless you really know what you are doing.
Disable autoscale at your own risk.
@end table
@section Advanced Video options
@@ -953,6 +617,8 @@ as the input (or graph output) and automatic conversions are disabled.
@item -sws_flags @var{flags} (@emph{input/output})
Set SwScaler flags.
@item -vdt @var{n}
Discard threshold.
@item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
Rate control override for specific intervals, formatted as "int,int,int"
@@ -964,8 +630,8 @@ factor if negative.
Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
Use this option if your input file is interlaced and you want
to keep the interlaced format for minimum losses.
The alternative is to deinterlace the input stream by use of a filter
such as @code{yadif} or @code{bwdif}, but deinterlacing introduces losses.
The alternative is to deinterlace the input stream with
@option{-deinterlace}, but deinterlacing introduces losses.
@item -psnr
Calculate PSNR of compressed frames.
@item -vstats
@@ -995,19 +661,12 @@ Deprecated see -bsf
@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
Force key frames at the specified timestamps, more precisely at the first
frames after each specified time.
@var{force_key_frames} can take arguments of the following form:
@table @option
@item @var{time}[,@var{time}...]
If the argument consists of timestamps, ffmpeg will round the specified times to the nearest
output timestamp as per the encoder time base and force a keyframe at the first frame having
timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too
coarse, then the keyframes may be forced on frames with timestamps lower than the specified time.
The default encoder time base is the inverse of the output framerate but may be set otherwise
via @code{-enc_time_base}.
If the argument is prefixed with @code{expr:}, the string @var{expr}
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
the time of the beginning of all chapters in the file, shifted by
@@ -1021,11 +680,6 @@ before the beginning of every chapter:
-force_key_frames 0:05:00,chapters-0.1
@end example
@item expr:@var{expr}
If the argument is prefixed with @code{expr:}, the string @var{expr}
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
The expression in @var{expr} can contain the following constants:
@table @option
@item n
@@ -1053,12 +707,6 @@ starting from second 13:
-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
@end example
@item source
If the argument is @code{source}, ffmpeg will force a key frame if
the current frame being encoded is marked as a key frame in its source.
@end table
Note that forcing too many keyframes is very harmful for the lookahead
algorithms of certain encoders: using fixed-GOP options or similar
would be more efficient.
@@ -1108,72 +756,6 @@ If not specified, @samp{auto_any} is used.
platform-appropriate subdevice (@samp{dxva2} or @samp{vaapi}) and then deriving a
QSV device from that.)
@item opencl
@var{device} selects the platform and device as @emph{platform_index.device_index}.
The set of devices can also be filtered using the key-value pairs to find only
devices matching particular platform or device strings.
The strings usable as filters are:
@table @option
@item platform_profile
@item platform_version
@item platform_name
@item platform_vendor
@item platform_extensions
@item device_name
@item device_vendor
@item driver_version
@item device_version
@item device_profile
@item device_extensions
@item device_type
@end table
The indices and filters must together uniquely select a device.
Examples:
@table @emph
@item -init_hw_device opencl:0.1
Choose the second device on the first platform.
@item -init_hw_device opencl:,device_name=Foo9000
Choose the device with a name containing the string @emph{Foo9000}.
@item -init_hw_device opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
Choose the GPU device on the second platform supporting the @emph{cl_khr_fp16}
extension.
@end table
@item vulkan
If @var{device} is an integer, it selects the device by its index in a
system-dependent list of devices. If @var{device} is any other string, it
selects the first device with a name containing that string as a substring.
The following options are recognized:
@table @option
@item debug
If set to 1, enables the validation layer, if installed.
@item linear_images
If set to 1, images allocated by the hwcontext will be linear and locally mappable.
@item instance_extensions
A plus separated list of additional instance extensions to enable.
@item device_extensions
A plus separated list of additional device extensions to enable.
@end table
Examples:
@table @emph
@item -init_hw_device vulkan:1
Choose the second device on the system.
@item -init_hw_device vulkan:RADV
Choose the first device with a name containing the string @emph{RADV}.
@item -init_hw_device vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface
Choose the first device and enable the Wayland and XCB instance extensions.
@end table
@end table
@item -init_hw_device @var{type}[=@var{name}]@@@var{source}
@@ -1204,6 +786,9 @@ Do not use any hardware acceleration (the default).
@item auto
Automatically select the hardware acceleration method.
@item vda
Use Apple VDA hardware acceleration.
@item vdpau
Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
@@ -1265,14 +850,8 @@ Set the number of audio channels. For output streams it is set by
default to the number of input audio channels. For input streams
this option only makes sense for audio grabbing devices and raw demuxers
and is mapped to the corresponding demuxer options.
@item -an (@emph{input/output})
As an input option, blocks all audio streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables audio recording i.e. automatic selection or
mapping of any audio stream. For full manual control see the @code{-map}
option.
@item -an (@emph{output})
Disable audio recording.
@item -acodec @var{codec} (@emph{input/output})
Set the audio codec. This is an alias for @code{-codec:a}.
@item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
@@ -1306,14 +885,8 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@table @option
@item -scodec @var{codec} (@emph{input/output})
Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{input/output})
As an input option, blocks all subtitle streams of a file from being filtered or
being automatically selected or mapped for any output. See @code{-discard}
option to disable streams individually.
As an output option, disables subtitle recording i.e. automatic selection or
mapping of any subtitle stream. For full manual control see the @code{-map}
option.
@item -sn (@emph{output})
Disable subtitle recording.
@item -sbsf @var{bitstream_filter}
Deprecated, see -bsf
@end table
@@ -1539,14 +1112,14 @@ disable any chapter copying.
@item -benchmark (@emph{global})
Show benchmarking information at the end of an encode.
Shows real, system and user time used and maximum memory consumption.
Shows CPU time used and maximum memory consumption.
Maximum memory consumption is not supported on all systems,
it will usually display as 0 if not supported.
@item -benchmark_all (@emph{global})
Show benchmarking information during the encode.
Shows real, system and user time used in various steps (audio/video encode/decode).
Shows CPU time used in various steps (audio/video encode/decode).
@item -timelimit @var{duration} (@emph{global})
Exit after ffmpeg has been running for @var{duration} seconds in CPU user time.
Exit after ffmpeg has been running for @var{duration} seconds.
@item -dump (@emph{global})
Dump each input packet to stderr.
@item -hex (@emph{global})
@@ -1559,6 +1132,14 @@ loss).
By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
Loop over the input stream. Currently it works only for image
streams. This option is used for automatic FFserver testing.
This option is deprecated, use -loop 1.
@item -loop_output @var{number_of_times}
Repeatedly loop output for formats that support looping such as animated GIF
(0 will loop the output infinitely).
This option is deprecated, use -loop.
@item -vsync @var{parameter}
Video sync method.
For compatibility reasons old values can be specified as numbers.
@@ -1608,17 +1189,6 @@ is enabled.
This option has been deprecated. Use the @code{aresample} audio filter instead.
@item -adrift_threshold @var{time}
Set the minimum difference between timestamps and audio data (in seconds) to trigger
adding/dropping samples to make it match the timestamps. This option effectively is
a threshold to select between hard (add/drop) and soft (squeeze/stretch) compensation.
@code{-async} must be set to a positive value.
@item -apad @var{parameters} (@emph{output,per-stream})
Pad the output audio stream(s). This is the same as applying @code{-af apad}.
Argument is a string of filter parameters composed the same as with the @code{apad} filter.
@code{-shortest} must be set for this output for the option to take effect.
@item -copyts
Do not process input timestamps, but keep their values without trying
to sanitize them. In particular, do not remove the initial start time
@@ -1683,19 +1253,13 @@ or as a floating point number (e.g. 0.04166, 2.0833e-5)
Default value is 0.
@item -bitexact (@emph{input/output})
Enable bitexact mode for (de)muxer and (de/en)coder
@item -shortest (@emph{output})
Finish encoding when the shortest input stream ends.
@item -dts_delta_threshold
Timestamp discontinuity delta threshold.
@item -dts_error_threshold @var{seconds}
Timestamp error delta threshold. This threshold use to discard crazy/damaged
timestamps and the default is 30 hours which is arbitrarily picked and quite
conservative.
@item -muxdelay @var{seconds} (@emph{output})
@item -muxdelay @var{seconds} (@emph{input})
Set the maximum demux-decode delay.
@item -muxpreload @var{seconds} (@emph{output})
@item -muxpreload @var{seconds} (@emph{input})
Set the initial demux-decode delay.
@item -streamid @var{output-stream-index}:@var{new-value} (@emph{output})
Assign a new stream-id value to an output stream. This option should be
@@ -1773,22 +1337,6 @@ graph will be added to the output file automatically, so we can simply write
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
@end example
As a special exception, you can use a bitmap subtitle stream as input: it
will be converted into a video with the same size as the largest video in
the file, or 720x576 if no video is present. Note that this is an
experimental and temporary solution. It will be removed once libavfilter has
proper support for subtitles.
For example, to hardcode subtitles on top of a DVB-T recording stored in
MPEG-TS format, delaying the subtitles by 1 second:
@example
ffmpeg -i input.ts -filter_complex \
'[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-sn -map '#0x2dc' output.mkv
@end example
(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
To generate 5 seconds of pure red video using lavfi @code{color} source:
@example
ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@@ -1824,9 +1372,18 @@ not start from timestamp 0, such as transport streams.
@item -thread_queue_size @var{size} (@emph{input})
This option sets the maximum number of queued packets when reading from the
file or device. With low latency / high rate live streams, packets may be
discarded if they are not read in a timely manner; setting this value can
force ffmpeg to use a separate input thread and read packets as soon as they
arrive. By default ffmpeg only do this if multiple inputs are specified.
discarded if they are not read in a timely manner; raising this value can
avoid it.
@item -override_ffserver (@emph{global})
Overrides the input specifications from @command{ffserver}. Using this
option you can map any input stream to @command{ffserver} and control
many aspects of the encoding from @command{ffmpeg}. Without this
option @command{ffmpeg} will transmit to @command{ffserver} what is
requested by @command{ffserver}.
The option is intended for cases where features are needed that cannot be
specified to @command{ffserver} but can be to @command{ffmpeg}.
@item -sdp_file @var{file} (@emph{global})
Print sdp information for an output stream to @var{file}.
@@ -1834,10 +1391,8 @@ This allows dumping sdp information when at least one output isn't an
rtp stream. (Requires at least one of the output formats to be rtp).
@item -discard (@emph{input})
Allows discarding specific streams or frames from streams.
Any input stream can be fully discarded, using value @code{all} whereas
selective discarding of frames from a stream occurs at the demuxer
and is not supported by all demuxers.
Allows discarding specific streams or frames of streams at the demuxer.
Not all demuxers support this.
@table @option
@item none
@@ -1865,15 +1420,8 @@ Stop and abort on various conditions. The following flags are available:
@table @option
@item empty_output
No packets were passed to the muxer, the output is empty.
@item empty_output_stream
No packets were passed to the muxer in some of the output streams.
@end table
@item -max_error_rate (@emph{global})
Set fraction of decoding frame failures across all inputs which when crossed
ffmpeg will return exit code 69. Crossing this threshold does not terminate
processing. Range is a floating-point number between 0 to 1. Default is 2/3.
@item -xerror (@emph{global})
Stop and exit on error
@@ -1886,23 +1434,24 @@ this buffer, in packets, for the matching output stream.
The default value of this option should be high enough for most uses, so only
touch this option if you are sure that you need it.
@item -muxing_queue_data_threshold @var{bytes} (@emph{output,per-stream})
This is a minimum threshold until which the muxing queue size is not taken into
account. Defaults to 50 megabytes per stream, and is based on the overall size
of packets passed to the muxer.
@item -auto_conversion_filters (@emph{global})
Enable automatically inserting format conversion filters in all filter
graphs, including those defined by @option{-vf}, @option{-af},
@option{-filter_complex} and @option{-lavfi}. If filter format negotiation
requires a conversion, the initialization of the filters will fail.
Conversions can still be performed by inserting the relevant conversion
filter (scale, aresample) in the graph.
On by default, to explicitly disable it you need to specify
@code{-noauto_conversion_filters}.
@end table
As a special exception, you can use a bitmap subtitle stream as input: it
will be converted into a video with the same size as the largest video in
the file, or 720x576 if no video is present. Note that this is an
experimental and temporary solution. It will be removed once libavfilter has
proper support for subtitles.
For example, to hardcode subtitles on top of a DVB-T recording stored in
MPEG-TS format, delaying the subtitles by 1 second:
@example
ffmpeg -i input.ts -filter_complex \
'[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
-sn -map '#0x2dc' output.mkv
@end example
(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
@section Preset files
A preset file contains a sequence of @var{option}=@var{value} pairs,
one for each line, specifying a sequence of options which would be
@@ -2179,7 +1728,6 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@ifset config-avfilter
@include filters.texi
@end ifset
@include general_contents.texi
@end ifset
@chapter See Also
@@ -2191,7 +1739,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@ifset config-not-all
@url{ffmpeg-all.html,ffmpeg-all},
@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -2210,7 +1758,7 @@ ffmpeg(1),
@ifset config-not-all
ffmpeg-all(1),
@end ifset
ffplay(1), ffprobe(1),
ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -60,14 +60,10 @@ Play @var{duration} seconds of audio/video.
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
Seek by bytes.
@item -seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
@item -nodisp
Disable graphical display.
@item -noborder
Borderless window.
@item -alwaysontop
Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
@item -volume
Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
@@ -76,10 +72,6 @@ as 100.
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -left @var{title}
Set the x position for the left of the window (default is a centered window).
@item -top @var{title}
Set the y position for the top of the window (default is a centered window).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
@@ -133,9 +125,8 @@ This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is shown by default, unless the
log level is lower than @code{info}. Its display can be forced by manually
specifying this option. To disable it, you need to specify @code{-nostats}.
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify @code{-nostats}.
@item -fast
Non-spec-compliant optimizations.
@@ -198,12 +189,6 @@ input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
@item -filter_threads @var{nb_threads}
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
@end table
@section While playing
@@ -295,7 +280,6 @@ Toggle full screen.
@ifset config-avfilter
@include filters.texi
@end ifset
@include general_contents.texi
@end ifset
@chapter See Also
@@ -307,7 +291,7 @@ Toggle full screen.
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -326,7 +310,7 @@ ffplay(1),
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1),
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -425,7 +425,7 @@ The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
Each section is printed on a single line.
If no option is specified, the output has the form:
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
@@ -584,14 +584,14 @@ value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_strict, x
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{https://www.w3.org/XML/}.
@url{http://www.w3.org/XML/}.
@c man end WRITERS
@chapter Timecode
@@ -642,7 +642,6 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-avfilter
@include filters.texi
@end ifset
@include general_contents.texi
@end ifset
@chapter See Also
@@ -654,7 +653,7 @@ DV, GXF and AVI timecodes are available in format metadata
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -673,7 +672,7 @@ ffprobe(1),
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1),
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)

View File

@@ -61,6 +61,8 @@
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="convergence_duration" type="xsd:long" />
<xsd:attribute name="convergence_duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
@@ -145,25 +147,11 @@
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:sequence>
<xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeList">
<xsd:sequence>
<xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeType">
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
@@ -213,6 +201,7 @@
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
@@ -223,7 +212,6 @@
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="closed_captions" type="xsd:boolean"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -235,6 +223,7 @@
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
@@ -354,6 +343,7 @@
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>

372
doc/ffserver.conf Normal file
View File

@@ -0,0 +1,372 @@
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
HTTPPort 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
HTTPBindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<Feed feed1.ffm>
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
# ffmpeg http://localhost:8090/feed1.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</Feed>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
<Stream test1.mpg>
# coming from live feed 'feed1'
Feed feed1.ffm
# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# mjpeg : Generate a M-JPEG stream.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg
# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32
# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 1
# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100
# Bitrate for the video stream
VideoBitRate 64
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize 160x128
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
#ACL ALLOW <first address> <last address>
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address> <last address>
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</Stream>
##################################################################
# Example streams
# Multipart JPEG
#<Stream test.mjpg>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</Stream>
# Single JPEG
#<Stream test.jpg>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</Stream>
# Flash
#<Stream test.swf>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</Stream>
# ASF compatible
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
# MP3 audio
#<Stream test.mp3>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Real with audio only at 32 kbits
#<Stream test.ra>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</Stream>
# Real with audio and video at 64 kbits
#<Stream test.rm>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</Stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<Stream file.rm>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</Stream>
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Metadata author "Me"
#Metadata copyright "Super MegaCorp"
#Metadata title "Test stream from disk"
#Metadata comment "Test comment"
#</Stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<Stream test1-rtsp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</Stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec aac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</Stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<Stream test1-sdp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</Stream>
##################################################################
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

923
doc/ffserver.texi Normal file
View File

@@ -0,0 +1,923 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION
@command{ffserver} is a streaming server for both audio and video.
It supports several live feeds, streaming from files and time shifting
on live feeds. You can seek to positions in the past on each live
feed, provided you specify a big enough feed storage.
@command{ffserver} is configured through a configuration file, which
is read at startup. If not explicitly specified, it will read from
@file{/etc/ffserver.conf}.
@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
RTP/RTSP/HTTP.
An @command{ffserver} instance will listen on some port as specified
in the configuration file. You can launch one or more instances of
@command{ffmpeg} and send one or more FFM streams to the port where
ffserver is expecting to receive them. Alternately, you can make
@command{ffserver} launch such @command{ffmpeg} instances at startup.
Input streams are called feeds, and each one is specified by a
@code{<Feed>} section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a @code{<Stream>} section in the
configuration file.
@chapter Detailed description
@command{ffserver} works by forwarding streams encoded by
@command{ffmpeg}, or pre-recorded streams which are read from disk.
Precisely, @command{ffserver} acts as an HTTP server, accepting POST
requests from @command{ffmpeg} to acquire the stream to publish, and
serving RTSP clients or HTTP clients GET requests with the stream
media content.
A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
a port where @command{ffserver} is listening.
Each feed is identified by a unique name, corresponding to the name
of the resource published on @command{ffserver}, and is configured by
a dedicated @code{Feed} section in the configuration file.
The feed publish URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
@end example
where @var{ffserver_ip_address} is the IP address of the machine where
@command{ffserver} is installed, @var{http_port} is the port number of
the HTTP server (configured through the @option{HTTPPort} option), and
@var{feed_name} is the name of the corresponding feed defined in the
configuration file.
Each feed is associated to a file which is stored on disk. This stored
file is used to send pre-recorded data to a player as fast as
possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by
@command{ffserver}, and made accessible through the HTTP protocol to
clients.
A stream can be connected to a feed, or to a file. In the first case,
the published stream is forwarded from the corresponding feed
generated by a running instance of @command{ffmpeg}, in the second
case the stream is read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name
of the resource served by @command{ffserver}, and is configured by
a dedicated @code{Stream} section in the configuration file.
The stream access HTTP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
@end example
The stream access RTSP URL is given by:
@example
http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
@end example
@var{stream_name} is the name of the corresponding stream defined in
the configuration file. @var{options} is a list of options specified
after the URL which affects how the stream is served by
@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
and RTSP ports configured with the options @var{HTTPPort} and
@var{RTSPPort} respectively.
In case the stream is associated to a feed, the encoding parameters
must be configured in the stream configuration. They are sent to
@command{ffmpeg} when setting up the encoding. This allows
@command{ffserver} to define the encoding parameters used by
the @command{ffmpeg} encoders.
The @command{ffmpeg} @option{override_ffserver} commandline option
allows one to override the encoding parameters set by the server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following
graph:
@verbatim
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
@end verbatim
@anchor{FFM}
@section FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@section Status stream
@command{ffserver} supports an HTTP interface which exposes the
current status of the server.
Simply point your browser to the address of the special status stream
specified in the configuration file.
For example if you have:
@example
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
@end example
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section How do I make it work?
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@example
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example
At this point you should be able to go to your Windows machine and fire up
Windows Media Player (WMP). Go to Open URL and enter
@example
http://<linuxbox>:8090/test.asf
@end example
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
You should edit the @file{ffserver.conf} file to suit your needs (in
terms of frame rates etc). Then install @command{ffserver} and
@command{ffmpeg}, write a script to start them up, and off you go.
@section What else can it do?
You can replay video from .ffm files that was recorded earlier.
However, there are a number of caveats, including the fact that the
ffserver parameters must match the original parameters used to record the
file. If they do not, then ffserver deletes the file before recording into it.
(Now that I write this, it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and
there are a bunch more parameters that you cannot control. Post a message
to the mailing list if there are some 'must have' parameters. Look in
ffserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used
in browsers. These files are actually redirections to the underlying ASF
or RM file. The reason for this is that the browser often fetches the
entire file before starting up the external viewer. The redirection files
are very small and can be transferred quickly. [The stream itself is
often 'infinite' and thus the browser tries to download it and never
finishes.]
@section Tips
* When you connect to a live stream, most players (WMP, RA, etc) want to
buffer a certain number of seconds of material so that they can display the
signal continuously. However, ffserver (by default) starts sending data
in realtime. This means that there is a pause of a few seconds while the
buffering is being done by the player. The good news is that this can be
cured by adding a '?buffer=5' to the end of the URL. This means that the
stream should start 5 seconds in the past -- and so the first 5 seconds
of the stream are sent as fast as the network will allow. It will then
slow down to real time. This noticeably improves the startup experience.
You can also add a 'Preroll 15' statement into the ffserver.conf that will
add the 15 second prebuffering on all requests that do not otherwise
specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
grabbed is marginally less than the number that ought to be grabbed. This
means that the timestamp in the encoded data stream gets behind realtime.
This means that if you say 'Preroll 10', then when the stream gets 10
or more seconds behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are
handled.
@section Does the @code{?date=} stuff work.
Yes (subject to the limitation outlined above). Also note that whenever you
start ffserver, it deletes the ffm file (if any parameters have changed),
thus wiping out what you had recorded before.
The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
of the following formats (the 'T' is literal):
@example
* YYYY-MM-DDTHH:MM:SS (localtime)
* YYYY-MM-DDTHH:MM:SSZ (UTC)
@end example
You can omit the YYYY-MM-DD, and then it refers to the current day. However
note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{configfile}
Read configuration file @file{configfile}. If not specified it will
read by default from @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the @code{Launch}
directives within the various @code{<Feed>} sections. Since
@command{ffserver} will not launch any @command{ffmpeg} instances, you
will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, and directs
log messages to stdout. When specified, the @option{CustomLog} option
is ignored.
@end table
@chapter Configuration file syntax
@command{ffserver} reads a configuration file containing global
options and settings for each stream and feed.
The configuration file consists of global options and dedicated
sections, which must be introduced by "<@var{SECTION_NAME}
@var{ARGS}>" on a separate line and must be terminated by a line in
the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
Currently the following sections are recognized: @samp{Feed},
@samp{Stream}, @samp{Redirect}.
A line starting with @code{#} is ignored and treated as a comment.
Name of options and sections are case-insensitive.
@section ACL syntax
An ACL (Access Control List) specifies the address which are allowed
to access a given stream, or to write a given feed.
It accepts the following forms
@itemize
@item
Allow/deny access to @var{address}.
@example
ACL ALLOW <address>
ACL DENY <address>
@end example
@item
Allow/deny access to ranges of addresses from @var{first_address} to
@var{last_address}.
@example
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
@end example
@end itemize
You can repeat the ACL allow/deny as often as you like. It is on a per
stream basis. The first match defines the action. If there are no matches,
then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost.
'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
allow everybody else.
@section Global options
@table @option
@item HTTPPort @var{port_number}
@item Port @var{port_number}
@item RTSPPort @var{port_number}
@var{HTTPPort} sets the HTTP server listening TCP port number,
@var{RTSPPort} sets the RTSP server listening TCP port number.
@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
You must select a different port from your standard HTTP web server if
it is running on the same computer.
If not specified, no corresponding server will be created.
@item HTTPBindAddress @var{ip_address}
@item BindAddress @var{ip_address}
@item RTSPBindAddress @var{ip_address}
Set address on which the HTTP/RTSP server is bound. Only useful if you
have several network interfaces.
@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
deprecated.
@item MaxHTTPConnections @var{n}
Set number of simultaneous HTTP connections that can be handled. It
has to be defined @emph{before} the @option{MaxClients} parameter,
since it defines the @option{MaxClients} maximum limit.
Default value is 2000.
@item MaxClients @var{n}
Set number of simultaneous requests that can be handled. Since
@command{ffserver} is very fast, it is more likely that you will want
to leave this high and use @option{MaxBandwidth}.
Default value is 5.
@item MaxBandwidth @var{kbps}
Set the maximum amount of kbit/sec that you are prepared to consume
when streaming to clients.
Default value is 1000.
@item CustomLog @var{filename}
Set access log file (uses standard Apache log file format). '-' is the
standard output.
If not specified @command{ffserver} will produce no log.
In case the commandline option @option{-d} is specified this option is
ignored, and the log is written to standard output.
@item NoDaemon
Set no-daemon mode. This option is currently ignored since now
@command{ffserver} will always work in no-daemon mode, and is
deprecated.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the all streams or not.
Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
The last occurrence overrides the previous if multiple definitions exist.
@end table
@section Feed section
A Feed section defines a feed provided to @command{ffserver}.
Each live feed contains one video and/or audio sequence coming from an
@command{ffmpeg} encoder or another @command{ffserver}. This sequence
may be encoded simultaneously with several codecs at several
resolutions.
A feed instance specification is introduced by a line in the form:
@example
<Feed FEED_FILENAME>
@end example
where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
@table @option
@item File @var{filename}
@item ReadOnlyFile @var{filename}
Set the path where the feed file is stored on disk.
If not specified, the @file{/tmp/FEED.ffm} is assumed, where
@var{FEED} is the feed name.
If @option{ReadOnlyFile} is used the file is marked as read-only and
it will not be deleted or updated.
@item Truncate
Truncate the feed file, rather than appending to it. By default
@command{ffserver} will append data to the file, until the maximum
file size value is reached (see @option{FileMaxSize} option).
@item FileMaxSize @var{size}
Set maximum size of the feed file in bytes. 0 means unlimited. The
postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
recognized.
Default value is 5M.
@item Launch @var{args}
Launch an @command{ffmpeg} command when creating @command{ffserver}.
@var{args} must be a sequence of arguments to be provided to an
@command{ffmpeg} instance. The first provided argument is ignored, and
it is replaced by a path with the same dirname of the @command{ffserver}
instance, followed by the remaining argument and terminated with a
path corresponding to the feed.
When the launched process exits, @command{ffserver} will launch
another program instance.
In case you need a more complex @command{ffmpeg} configuration,
e.g. if you need to generate multiple FFM feeds with a single
@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
This option is ignored in case the commandline option @option{-n} is
specified.
@item ACL @var{spec}
Specify the list of IP address which are allowed or denied to write
the feed. Multiple ACL options can be specified.
@end table
@section Stream section
A Stream section defines a stream provided by @command{ffserver}, and
identified by a single name.
The stream is sent when answering a request containing the stream
name.
A stream section must be introduced by the line:
@example
<Stream STREAM_NAME>
@end example
where @var{STREAM_NAME} specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the @emph{encoding} tag, and they are
used to set the encoding parameters, and are mapped to libavcodec
encoding options. Not all encoding options are supported, in
particular it is not possible to set encoder private options. In order
to override the encoding options specified by @command{ffserver}, you
can use the @command{ffmpeg} @option{override_ffserver} commandline
option.
Only one of the @option{Feed} and @option{File} options should be set.
@table @option
@item Feed @var{feed_name}
Set the input feed. @var{feed_name} must correspond to an existing
feed defined in a @code{Feed} section.
When this option is set, encoding options are used to setup the
encoding operated by the remote @command{ffmpeg} process.
@item File @var{filename}
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input
file content is re-streamed as is.
@item Format @var{format_name}
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to
@samp{status}, it is treated as a status stream.
@item InputFormat @var{format_name}
Set input format. If not specified, it is automatically guessed.
@item Preroll @var{n}
Set this to the number of seconds backwards in time to start. Note that
most players will buffer 5-10 seconds of video, and also you need to allow
for a keyframe to appear in the data stream.
Default value is 0.
@item StartSendOnKey
Do not send stream until it gets the first key frame. By default
@command{ffserver} will send data immediately.
@item MaxTime @var{n}
Set the number of seconds to run. This value set the maximum duration
of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
@item ACL @var{spec}
Set ACL for the stream.
@item DynamicACL @var{spec}
@item RTSPOption @var{option}
@item MulticastAddress @var{address}
@item MulticastPort @var{port}
@item MulticastTTL @var{integer}
@item NoLoop
@item FaviconURL @var{url}
Set favicon (favourite icon) for the server status page. It is ignored
for regular streams.
@item Author @var{value}
@item Comment @var{value}
@item Copyright @var{value}
@item Title @var{value}
Set metadata corresponding to the option. All these options are
deprecated in favor of @option{Metadata}.
@item Metadata @var{key} @var{value}
Set metadata value on the output stream.
@item UseDefaults
@item NoDefaults
Control whether default codec options are used for the stream or not.
Default is @var{UseDefaults} unless disabled globally.
@item NoAudio
@item NoVideo
Suppress audio/video.
@item AudioCodec @var{codec_name} (@emph{encoding,audio})
Set audio codec.
@item AudioBitRate @var{rate} (@emph{encoding,audio})
Set bitrate for the audio stream in kbits per second.
@item AudioChannels @var{n} (@emph{encoding,audio})
Set number of audio channels.
@item AudioSampleRate @var{n} (@emph{encoding,audio})
Set sampling frequency for audio. When using low bitrates, you should
lower this frequency to 22050 or 11025. The supported frequencies
depend on the selected audio codec.
@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
Set generic or private option for audio stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVPresetAudio @var{preset} (@emph{encoding,audio})
Set preset for audio stream.
@item VideoCodec @var{codec_name} (@emph{encoding,video})
Set video codec.
@item VideoBitRate @var{n} (@emph{encoding,video})
Set bitrate for the video stream in kbits per second.
@item VideoBitRateRange @var{range} (@emph{encoding,video})
Set video bitrate range.
A range must be specified in the form @var{minrate}-@var{maxrate}, and
specifies the @option{minrate} and @option{maxrate} encoding options
expressed in kbits per second.
@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
Set video bitrate tolerance in kbits per second.
@item PixelFormat @var{pixel_format} (@emph{encoding,video})
Set video pixel format.
@item Debug @var{integer} (@emph{encoding,video})
Set video @option{debug} encoding option.
@item Strict @var{integer} (@emph{encoding,video})
Set video @option{strict} encoding option.
@item VideoBufferSize @var{n} (@emph{encoding,video})
Set ratecontrol buffer size, expressed in KB.
@item VideoFrameRate @var{n} (@emph{encoding,video})
Set number of video frames per second.
@item VideoSize (@emph{encoding,video})
Set size of the video frame, must be an abbreviation or in the form
@var{W}x@var{H}. See @ref{video size syntax,,the Video size section
in the ffmpeg-utils(1) manual,ffmpeg-utils}.
Default value is @code{160x128}.
@item VideoIntraOnly (@emph{encoding,video})
Transmit only intra frames (useful for low bitrates, but kills frame rate).
@item VideoGopSize @var{n} (@emph{encoding,video})
If non-intra only, an intra frame is transmitted every VideoGopSize
frames. Video synchronization can only begin at an intra frame.
@item VideoTag @var{tag} (@emph{encoding,video})
Set video tag.
@item VideoHighQuality (@emph{encoding,video})
@item Video4MotionVector (@emph{encoding,video})
@item BitExact (@emph{encoding,video})
Set bitexact encoding flag.
@item IdctSimple (@emph{encoding,video})
Set simple IDCT algorithm.
@item Qscale @var{n} (@emph{encoding,video})
Enable constant quality encoding, and set video qscale (quantization
scale) value, expressed in @var{n} QP units.
@item VideoQMin @var{n} (@emph{encoding,video})
@item VideoQMax @var{n} (@emph{encoding,video})
Set video qmin/qmax.
@item VideoQDiff @var{integer} (@emph{encoding,video})
Set video @option{qdiff} encoding option.
@item LumiMask @var{float} (@emph{encoding,video})
@item DarkMask @var{float} (@emph{encoding,video})
Set @option{lumi_mask}/@option{dark_mask} encoding options.
@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
Set generic or private option for video stream.
Private option must be prefixed with codec name or codec must be defined before.
@item AVPresetVideo @var{preset} (@emph{encoding,video})
Set preset for video stream.
@var{preset} must be the path of a preset file.
@end table
@subsection Server status stream
A server status stream is a special stream which is used to show
statistics about the @command{ffserver} operations.
It must be specified setting the option @option{Format} to
@samp{status}.
@section Redirect section
A redirect section specifies where to redirect the requested URL to
another page.
A redirect section must be introduced by the line:
@example
<Redirect NAME>
@end example
where @var{NAME} is the name of the page which should be redirected.
It only accepts the option @option{URL}, which specify the redirection
URL.
@chapter Stream examples
@itemize
@item
Multipart JPEG
@example
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
@end example
@item
Single JPEG
@example
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
@end example
@item
Flash
@example
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
@end example
@item
ASF compatible
@example
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
@end example
@item
MP3 audio
@example
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Ogg Vorbis audio
@example
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
@end example
@item
Real with audio only at 32 kbits
@example
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
@end example
@item
Real with audio and video at 64 kbits
@example
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
@end example
@item
For stream coming from a file: you only need to set the input filename
and optionally a new format.
@example
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
@end example
@example
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
@end example
@end itemize
@c man end
@include config.texi
@ifset config-all
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffserver.html,ffserver},
@end ifset
@ifset config-not-all
@url{ffserver-all.html,ffserver-all},
@end ifset
the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffserver(1),
@end ifset
@ifset config-not-all
ffserver-all(1),
@end ifset
the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@end ignore
@bye

View File

@@ -34,25 +34,18 @@ Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4. If @var{stream_index} is used as an
additional stream specifier (see below), then it selects stream number
@var{stream_index} from the matching streams. Stream numbering is based on the
order of the streams as detected by libavformat except when a program ID is
also specified. In this case it is based on the ordering of the streams in the
program.
@item @var{stream_type}[:@var{additional_stream_specifier}]
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
streams, 'V' only matches video streams which are not attached pictures, video
thumbnails or cover arts. If @var{additional_stream_specifier} is used, then
it matches streams which both have this type and match the
@var{additional_stream_specifier}. Otherwise, it matches all streams of the
specified type.
@item p:@var{program_id}[:@var{additional_stream_specifier}]
Matches streams which are in the program with the id @var{program_id}. If
@var{additional_stream_specifier} is used, then it matches streams which both
are part of the program and match the @var{additional_stream_specifier}.
thumbnails or cover arts. If @var{stream_index} is given, then it matches
stream number @var{stream_index} of this type. Otherwise, it matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
program.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
@@ -107,24 +100,13 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@item filter=@var{filter_name}
Print detailed information about the filter named @var{filter_name}. Use the
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@item bsf=@var{bitstream_filter_name}
Print detailed information about the bitstream filter named @var{bitstream_filter_name}.
Use the @option{-bsfs} option to get a list of all bitstream filters.
@item protocol=@var{protocol_name}
Print detailed information about the protocol named @var{protocol_name}.
Use the @option{-protocols} option to get a list of all protocols.
@end table
@item -version
Show version.
@item -buildconf
Show the build configuration, one option per line.
@item -formats
Show available formats (including devices).
@@ -186,24 +168,14 @@ The returned list cannot be assumed to be always complete.
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [@var{flags}+]@var{loglevel} | -v [@var{flags}+]@var{loglevel}
Set logging level and flags used by the library.
The optional @var{flags} prefix can consist of the following values:
@table @samp
@item repeat
Indicates that repeated log output should not be compressed to the first line
and the "Last message repeated n times" line will be omitted.
@item level
Indicates that log output should add a @code{[level]} prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
@end table
Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
flag without affecting other @var{flags} or changing @var{loglevel}. When
setting both @var{flags} and @var{loglevel}, a '+' separator is expected
between the last @var{flags} value and before @var{loglevel}.
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
Set the logging level used by the library.
Adding "repeat+" indicates that repeated log output should not be compressed
to the first line and the "Last message repeated n times" line will be
omitted. "repeat" can also be used alone.
If "repeat" is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
'repeat' will not change the loglevel.
@var{loglevel} is a string or a number containing one of the following values:
@table @samp
@item quiet, -8
@@ -229,29 +201,20 @@ Show everything, including debugging information.
@item trace, 56
@end table
For example to enable repeated log output, add the @code{level} prefix, and set
@var{loglevel} to @code{verbose}:
@example
ffmpeg -loglevel repeat+level+verbose -i input output
@end example
Another example that enables repeated log output without affecting current
state of @code{level} prefix flag or @var{loglevel}:
@example
ffmpeg [...] -loglevel +repeat
@end example
By default the program logs to stderr. If coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR}, or can be forced setting
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a future FFmpeg version.
@item -report
Dump full command line and log output to a file named
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel debug}.
It also implies @code{-loglevel verbose}.
Setting the environment variable @env{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
@@ -353,11 +316,50 @@ Possible flags for this option are:
@end table
@end table
@item -max_alloc @var{bytes}
Set the maximum size limit for allocating a block on the heap by ffmpeg's
family of malloc functions. Exercise @strong{extreme caution} when using
this option. Don't use if you do not understand the full consequence of doing so.
Default is INT_MAX.
@item -opencl_bench
This option is used to benchmark all available OpenCL devices and print the
results. This option is only available when FFmpeg has been compiled with
@code{--enable-opencl}.
When FFmpeg is configured with @code{--enable-opencl}, the options for the
global OpenCL context are set via @option{-opencl_options}. See the
"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
supported options. Amongst others, these options include the ability to select
a specific platform and device to run the OpenCL code on. By default, FFmpeg
will run on the first device of the first platform. While the options for the
global OpenCL context provide flexibility to the user in selecting the OpenCL
device of their choice, most users would probably want to select the fastest
OpenCL device for their system.
This option assists the selection of the most efficient configuration by
identifying the appropriate device for the user's system. The built-in
benchmark is run on all the OpenCL devices and the performance is measured for
each device. The devices in the results list are sorted based on their
performance with the fastest device listed first. The user can subsequently
invoke @command{ffmpeg} using the device deemed most appropriate via
@option{-opencl_options} to obtain the best performance for the OpenCL
accelerated code.
Typical usage to use the fastest OpenCL device involve the following steps.
Run the command:
@example
ffmpeg -opencl_bench
@end example
Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
i.e. fastest device in the list.
Select the platform and device using the command:
@example
ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
@end example
@item -opencl_options options (@emph{global})
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with @code{--enable-opencl}.
@var{options} must be a list of @var{key}=@var{value} option pairs
separated by ':'. See the ``OpenCL Options'' section in the
ffmpeg-utils manual for the list of supported options.
@end table
@section AVOptions
@@ -383,15 +385,7 @@ ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them:
@example
ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
@end example
In the above example, a multichannel audio stream is mapped twice for output.
The first instance is encoded with codec ac3 and bitrate 640k.
The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using
absolute index of the output stream.
should be attached to them.
Note: the @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.

File diff suppressed because it is too large Load Diff

View File

@@ -27,50 +27,40 @@ stream information. A higher value will enable detecting more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item max_probe_packets @var{integer} (@emph{input})
Set the maximum number of buffered packets when probing a codec.
Default is 2500 packets.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags}
Set format flags. Some are implemented for a limited number of formats.
@item fflags @var{flags} (@emph{input/output})
Set format flags.
Possible values for input files:
Possible values:
@table @samp
@item discardcorrupt
Discard corrupted packets.
@item ignidx
Ignore index.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate missing PTS if DTS is present.
@item igndts
Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
Ignore index.
@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.
Generate PTS.
@item nofillin
Do not fill in missing values in packet fields that can be exactly calculated.
Do not fill in missing values that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item igndts
Ignore DTS.
@item discardcorrupt
Discard corrupted frames.
@item sortdts
Try to interleave output packets by DTS. At present, available only for AVIs with an index.
@end table
Possible values for output files:
@table @samp
@item autobsf
Automatically apply bitstream filters as required by the output format. Enabled by default.
Try to interleave output packets by DTS.
@item keepside
Do not merge side data.
@item latm
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@item bitexact
Only write platform-, build- and time-independent data.
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item flush_packets
Write out packets immediately.
@item latm (@emph{deprecated},@emph{inert})
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flushing the longer
@@ -143,7 +133,7 @@ Consider things that a sane encoder should not do as an error.
@item max_interleave_delta @var{integer} (@emph{output})
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 10000000 (10 seconds).
expressed in microseconds, and defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat will
wait until it has at least one packet for each stream before actually
@@ -215,7 +205,7 @@ is @code{0} (meaning that no offset is applied).
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
For example to separate the fields with newlines and indention:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@@ -224,32 +214,6 @@ ffprobe -dump_separator "
@item max_streams @var{integer} (@emph{input})
Specifies the maximum number of streams. This can be used to reject files that
would require too many resources due to a large number of streams.
@item skip_estimate_duration_from_pts @var{bool} (@emph{input})
Skip estimation of input duration when calculated using PTS.
At present, applicable for MPEG-PS and MPEG-TS.
@item strict, f_strict @var{integer} (@emph{input/output})
Specify how strictly to follow the standards. @code{f_strict} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item very
strictly conform to an older more strict version of the spec or reference software
@item strict
strictly conform to all the things in the spec no matter what consequences
@item normal
@item unofficial
allow unofficial extensions
@item experimental
allow non standardized experimental things, experimental
(unfinished/work in progress/not well tested) decoders and encoders.
Note: experimental decoders can pose a security risk, do not use this for
decoding untrusted input.
@end table
@end table
@c man end FORMAT OPTIONS
@@ -260,10 +224,30 @@ decoding untrusted input.
Format stream specifiers allow selection of one or more streams that
match specific properties.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of following: 'v' for video, 'a' for audio,
's' for subtitle, 'd' for data, and 't' for attachments. If
@var{stream_index} is given, then it matches the stream number
@var{stream_index} of this type. Otherwise, it matches all streams of
this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then it matches the stream with number
@var{stream_index} in the program with the id
@var{program_id}. Otherwise, it matches all streams in the program.
@item #@var{stream_id}
Matches the stream by a format-specific ID.
@end table
The exact semantics of stream specifiers is defined by the
@code{avformat_match_stream_specifier()} function declared in the
@file{libavformat/avformat.h} header and documented in the
@ref{Stream specifiers,,Stream specifiers section in the ffmpeg(1) manual,ffmpeg}.
@file{libavformat/avformat.h} header.
@ifclear config-writeonly
@include demuxers.texi

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@@ -63,46 +63,6 @@ Set the number of channels. Default is 2.
@end table
@section android_camera
Android camera input device.
This input devices uses the Android Camera2 NDK API which is
available on devices with API level 24+. The availability of
android_camera is autodetected during configuration.
This device allows capturing from all cameras on an Android device,
which are integrated into the Camera2 NDK API.
The available cameras are enumerated internally and can be selected
with the @var{camera_index} parameter. The input file string is
discarded.
Generally the back facing camera has index 0 while the front facing
camera has index 1.
@subsection Options
@table @option
@item video_size
Set the video size given as a string such as 640x480 or hd720.
Falls back to the first available configuration reported by
Android if requested video size is not available or by default.
@item framerate
Set the video framerate.
Falls back to the first available configuration reported by
Android if requested framerate is not available or by default (-1).
@item camera_index
Set the index of the camera to use. Default is 0.
@item input_queue_size
Set the maximum number of frames to buffer. Default is 5.
@end table
@section avfoundation
AVFoundation input device.
@@ -178,9 +138,6 @@ Capture the mouse pointer. Default is 0.
@item -capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.
@item -capture_raw_data
Capture the raw device data. Default is 0.
Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.
@end table
@subsection Examples
@@ -211,13 +168,6 @@ Record video from the system default video device using the pixel format bgr0 an
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
@end example
@item
Record raw DV data from a suitable input device and write the output into out.dv:
@example
$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
@end example
@end itemize
@section bktr
@@ -277,8 +227,7 @@ audio track.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}. This option is deprecated, please use the
@code{-sources} option of ffmpeg to list the available input devices.
Defaults to @option{false}.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
@@ -289,38 +238,26 @@ This sets the input video format to the format given by the FourCC. To see
the supported values of your device(s) use @option{list_formats}.
Note that there is a FourCC @option{'pal '} that can also be used
as @option{pal} (3 letters).
Default behavior is autodetection of the input video format, if the hardware
supports it.
@item bm_v210
This is a deprecated option, you can use @option{raw_format} instead.
If set to @samp{1}, video is captured in 10 bit v210 instead
of uyvy422. Not all Blackmagic devices support this option.
@item raw_format
Set the pixel format of the captured video.
Available values are:
@table @samp
@item auto
This is the default which means 8-bit YUV 422 or 8-bit ARGB if format
autodetection is used, 8-bit YUV 422 otherwise.
@item uyvy422
8-bit YUV 422.
@item yuv422p10
10-bit YUV 422.
@item argb
8-bit RGB.
@item bgra
8-bit RGB.
@item rgb10
10-bit RGB.
@end table
@item teletext_lines
@@ -347,17 +284,6 @@ Defaults to @samp{2}.
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
@item timecode_format
Timecode type to include in the frame and video stream metadata. Must be
@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
@samp{rp188hfr}, @samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}.
Defaults to @samp{none} (not included).
In order to properly support 50/60 fps timecodes, the ordering of the queried
timecode types for @samp{rp188any} is HFR, VITC1, VITC2 and LTC for >30 fps
content. Note that this is slightly different to the ordering used by the
DeckLink API, which is HFR, VITC1, LTC, VITC2.
@item video_input
Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
@samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
@@ -370,13 +296,11 @@ Sets the audio input source. Must be @samp{unset}, @samp{embedded},
@item video_pts
Sets the video packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
Defaults to @samp{video}.
@samp{reference} or @samp{wallclock}. Defaults to @samp{video}.
@item audio_pts
Sets the audio packet timestamp source. Must be @samp{video}, @samp{audio},
@samp{reference}, @samp{wallclock} or @samp{abs_wallclock}.
Defaults to @samp{audio}.
@samp{reference} or @samp{wallclock}. Defaults to @samp{audio}.
@item draw_bars
If set to @samp{true}, color bars are drawn in the event of a signal loss.
@@ -387,43 +311,6 @@ Sets maximum input buffer size in bytes. If the buffering reaches this value,
incoming frames will be dropped.
Defaults to @samp{1073741824}.
@item audio_depth
Sets the audio sample bit depth. Must be @samp{16} or @samp{32}.
Defaults to @samp{16}.
@item decklink_copyts
If set to @option{true}, timestamps are forwarded as they are without removing
the initial offset.
Defaults to @option{false}.
@item timestamp_align
Capture start time alignment in seconds. If set to nonzero, input frames are
dropped till the system timestamp aligns with configured value.
Alignment difference of up to one frame duration is tolerated.
This is useful for maintaining input synchronization across N different
hardware devices deployed for 'N-way' redundancy. The system time of different
hardware devices should be synchronized with protocols such as NTP or PTP,
before using this option.
Note that this method is not foolproof. In some border cases input
synchronization may not happen due to thread scheduling jitters in the OS.
Either sync could go wrong by 1 frame or in a rarer case
@option{timestamp_align} seconds.
Defaults to @samp{0}.
@item wait_for_tc (@emph{bool})
Drop frames till a frame with timecode is received. Sometimes serial timecode
isn't received with the first input frame. If that happens, the stored stream
timecode will be inaccurate. If this option is set to @option{true}, input frames
are dropped till a frame with timecode is received.
Option @var{timecode_format} must be specified.
Defaults to @option{false}.
@item enable_klv(@emph{bool})
If set to @option{true}, extracts KLV data from VANC and outputs KLV packets.
KLV VANC packets are joined based on MID and PSC fields and aggregated into
one KLV packet.
Defaults to @option{false}.
@end table
@subsection Examples
@@ -433,7 +320,7 @@ Defaults to @option{false}.
@item
List input devices:
@example
ffmpeg -sources decklink
ffmpeg -f decklink -list_devices 1 -i dummy
@end example
@item
@@ -451,7 +338,7 @@ ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy outp
@item
Capture video clip at 1080i50 10 bit:
@example
ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
@end example
@item
@@ -462,6 +349,116 @@ ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder'
@end itemize
@section kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at
@option{x11grab} instead.
@subsection Options
@table @option
@item device
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. Defaults to @option{bgr0}.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs, but can't be autodetected. See the libdrm documentation for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
will be used.
@item plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither @option{crtc_id} nor @option{plane_id} are specified.
@item framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to @code{30}.
@end table
@subsection Examples
@itemize
@item
Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.
@example
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
@end example
@item
Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
@example
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
@end example
@end itemize
@section libndi_newtek
The libndi_newtek input device provides capture capabilities for using NDI (Network
Device Interface, standard created by NewTek).
Input filename is a NDI source name that could be found by sending -find_sources 1
to command line - it has no specific syntax but human-readable formatted.
To enable this input device, you need the NDI SDK and you
need to configure with the appropriate @code{--extra-cflags}
and @code{--extra-ldflags}.
@subsection Options
@table @option
@item find_sources
If set to @option{true}, print a list of found/available NDI sources and exit.
Defaults to @option{false}.
@item wait_sources
Override time to wait until the number of online sources have changed.
Defaults to @option{0.5}.
@item allow_video_fields
When this flag is @option{false}, all video that you receive will be progressive.
Defaults to @option{true}.
@end table
@subsection Examples
@itemize
@item
List input devices:
@example
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
@end example
@item
Restream to NDI:
@example
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
@end example
@end itemize
@section dshow
Windows DirectShow input device.
@@ -826,7 +823,7 @@ ffplay -f iec61883 -i auto
Grab and record the input of a FireWire DV/HDV device,
using a packet buffer of 100000 packets if the source is HDV.
@example
ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
@end example
@end itemize
@@ -889,80 +886,6 @@ Set the number of channels. Default is 2.
@end table
@section kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at
@option{x11grab} instead.
@subsection Options
@table @option
@item device
DRM device to capture on. Defaults to @option{/dev/dri/card0}.
@item format
Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7
or later, but needs to be provided for earlier versions. Defaults to @option{bgr0},
which is the most common format used by the Linux console and Xorg X server.
@item format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need
to be provided explicitly when needed in earlier versions. See the libdrm documentation
for possible values.
@item crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC
will be used.
@item plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither @option{crtc_id} nor @option{plane_id} are specified.
@item framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to @code{30}.
@end table
@subsection Examples
@itemize
@item
Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.
@example
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
@end example
@item
Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
@example
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
@end example
@item
To capture only part of a plane the output can be cropped - this can be used to capture
a single window, as long as it has a known absolute position and size. For example, to
capture and encode the middle quarter of a 1920x1080 plane:
@example
ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
@end example
@end itemize
@section lavfi
Libavfilter input virtual device.
@@ -1101,21 +1024,6 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option @code{--enable-libdc1394}.
@subsection Options
@table @option
@item framerate
Set the frame rate. Default is @code{ntsc}, corresponding to a frame
rate of @code{30000/1001}.
@item pixel_format
Select the pixel format. Default is @code{uyvy422}.
@item video_size
Set the video size given as a string such as @code{640x480} or @code{hd720}.
Default is @code{qvga}.
@end table
@section openal
The OpenAL input device provides audio capture on all systems with a
@@ -1234,6 +1142,7 @@ Set the number of channels. Default is 2.
@end table
@section pulse
PulseAudio input device.
@@ -1508,14 +1417,6 @@ ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@subsection Options
@table @option
@item select_region
Specify whether to select the grabbing area graphically using the pointer.
A value of @code{1} prompts the user to select the grabbing area graphically
by clicking and dragging. A single click with no dragging will select the
whole screen. A region with zero width or height will also select the whole
screen. This option overwrites the @var{video_size}, @var{grab_x}, and
@var{grab_y} options. Default value is @code{0}.
@item draw_mouse
Specify whether to draw the mouse pointer. A value of @code{0} specifies
not to draw the pointer. Default value is @code{1}.
@@ -1564,21 +1465,8 @@ With @var{follow_mouse}:
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example
@item window_id
Grab this window, instead of the whole screen. Default value is 0, which maps to
the whole screen (root window).
The id of a window can be found using the @command{xwininfo} program, possibly with options -tree and
-root.
If the window is later enlarged, the new area is not recorded. Video ends when
the window is closed, unmapped (i.e., iconified) or shrunk beyond the video
size (which defaults to the initial window size).
This option disables options @option{follow_mouse} and @option{select_region}.
@item video_size
Set the video frame size. Default is the full desktop or window.
Set the video frame size. Default value is @code{vga}.
@item grab_x
@item grab_y

View File

@@ -193,6 +193,9 @@ ffplay
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
postproc
issues in libpostproc/*

View File

@@ -94,17 +94,18 @@ Stuff that didn't reach the codebase:
- a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
- 0cef06df0 checkasm: add HEVC MC tests
- e7078e842 hevcdsp: add x86 SIMD for MC
- 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
- VAAPI VP8 decode hwaccel (currently under review: http://ffmpeg.org/pipermail/ffmpeg-devel/2017-February/thread.html#207348)
- Removal of the custom atomic API (5cc0057f49, see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-March/209003.html)
- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
- use of the bsf instead of our parser for vp9 superframes (see fa1749dd34)
- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
- f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
- 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
- Support recovery from an already present HLS playlist (see 16cb06bb30)
- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
Collateral damage that needs work locally:
------------------------------------------
- Merge proresdec2.c and proresdec_lgpl.c
- Merge proresenc_anatoliy.c and proresenc_kostya.c
- Fix MIPS AC3 downmix

View File

@@ -26,13 +26,13 @@ implementing robust and fast codecs as well as for experimentation.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
libavutil(3)
@end ifnothtml

View File

@@ -23,13 +23,13 @@ VfW, DShow, and ALSA.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-devices(1),
libavutil(3), libavcodec(3), libavformat(3)
@end ifnothtml

View File

@@ -21,14 +21,14 @@ framework containing several filters, sources and sinks.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-filters.html,ffmpeg-filters},
@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-filters(1),
libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
@end ifnothtml

View File

@@ -26,13 +26,13 @@ resource.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-formats(1), ffmpeg-protocols(1),
libavutil(3), libavcodec(3)
@end ifnothtml

View File

@@ -42,12 +42,12 @@ It should avoid useless features that almost no one needs.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1)
@end ifnothtml

View File

@@ -48,13 +48,13 @@ enabled through dedicated options.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-resampler(1),
libavutil(3)
@end ifnothtml

View File

@@ -41,13 +41,13 @@ colorspaces differ.
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-scaler(1),
libavutil(3)
@end ifnothtml

Some files were not shown because too many files have changed in this diff Show More