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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-12-03 02:08:55 +01:00
parent 12804348f5 d268b79e34
commit e4de71677f
87 changed files with 2226 additions and 1307 deletions
Download Patch File Download Diff File Expand all files Collapse all files

106
libavcodec/aacdec.c
View File

@@ -471,15 +471,17 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
* @param data pointer to AVCodecContext extradata
* @param data_size size of AVCCodecContext extradata
* @param data pointer to buffer holding an audio specific config
* @param bit_size size of audio specific config or data in bits
* @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
const uint8_t *data, int data_size, int asclen)
const uint8_t *data, int bit_size,
int sync_extension)
{
GetBitContext gb;
int i;
@@ -489,9 +491,9 @@ static int decode_audio_specific_config(AACContext *ac,
av_dlog(avctx, "%02x ", avctx->extradata[i]);
av_dlog(avctx, "\n");
init_get_bits(&gb, data, data_size * 8);
init_get_bits(&gb, data, bit_size);
if ((i = avpriv_mpeg4audio_get_config(m4ac, data, asclen/8)) < 0)
if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
return -1;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
@@ -591,7 +593,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
avctx->extradata,
avctx->extradata_size, 8*avctx->extradata_size) < 0)
avctx->extradata_size*8, 1) < 0)
return -1;
} else {
int sr, i;
@@ -665,6 +667,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
cbrt_tableinit();
avcodec_get_frame_defaults(&ac->frame);
avctx->coded_frame = &ac->frame;
return 0;
}
@@ -2132,12 +2137,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
int *data_size, GetBitContext *gb)
int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id, data_size_tmp;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
@@ -2250,24 +2255,26 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
avctx->frame_size = samples;
}
data_size_tmp = samples * avctx->channels *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
*data_size, data_size_tmp);
return -1;
}
*data_size = data_size_tmp;
if (samples) {
/* get output buffer */
ac->frame.nb_samples = samples;
if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return err;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
else
ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
*(AVFrame *)data = ac->frame;
}
*got_frame_ptr = !!samples;
if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
@@ -2276,7 +2283,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@@ -2287,7 +2294,7 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
init_get_bits(&gb, buf, buf_size * 8);
if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -2340,30 +2347,40 @@ static inline uint32_t latm_get_value(GetBitContext *b)
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
GetBitContext *gb, int asclen)
{
AVCodecContext *avctx = latmctx->aac_ctx.avctx;
AACContext *ac= &latmctx->aac_ctx;
MPEG4AudioConfig m4ac=ac->m4ac;
int config_start_bit = get_bits_count(gb);
int bits_consumed, esize;
AACContext *ac = &latmctx->aac_ctx;
AVCodecContext *avctx = ac->avctx;
MPEG4AudioConfig m4ac = {0};
int config_start_bit = get_bits_count(gb);
int sync_extension = 0;
int bits_consumed, esize;
if (asclen) {
sync_extension = 1;
asclen = FFMIN(asclen, get_bits_left(gb));
} else
asclen = get_bits_left(gb);
if (config_start_bit % 8) {
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
"config not byte aligned.\n", 1);
return AVERROR_INVALIDDATA;
} else {
bits_consumed =
decode_audio_specific_config(ac, avctx, &m4ac,
}
bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
get_bits_left(gb) / 8, asclen);
asclen, sync_extension);
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
if(ac->m4ac.sample_rate != m4ac.sample_rate || m4ac.chan_config != ac->m4ac.chan_config)
ac->m4ac= m4ac;
if (bits_consumed < 0)
return AVERROR_INVALIDDATA;
if (ac->m4ac.sample_rate != m4ac.sample_rate ||
ac->m4ac.chan_config != m4ac.chan_config) {
av_log(avctx, AV_LOG_INFO, "audio config changed\n");
latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
if (avctx->extradata_size <= esize) {
if (avctx->extradata_size < esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
@@ -2373,9 +2390,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
skip_bits_long(gb, bits_consumed);
}
skip_bits_long(gb, bits_consumed);
return bits_consumed;
}
@@ -2512,8 +2528,8 @@ static int read_audio_mux_element(struct LATMContext *latmctx,
}
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
AVPacket *avpkt)
static int latm_decode_frame(AVCodecContext *avctx, void *out,
int *got_frame_ptr, AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
@@ -2535,12 +2551,12 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
if (!latmctx->initialized) {
if (!avctx->extradata) {
*out_size = 0;
*got_frame_ptr = 0;
return avpkt->size;
} else {
if ((err = decode_audio_specific_config(
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac,
avctx->extradata, avctx->extradata_size, 8*avctx->extradata_size)) < 0)
avctx->extradata, avctx->extradata_size*8, 1)) < 0)
return err;
latmctx->initialized = 1;
}
@@ -2553,7 +2569,7 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
return AVERROR_INVALIDDATA;
}
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
return err;
return muxlength;
@@ -2583,7 +2599,7 @@ AVCodec ff_aac_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
@@ -2604,7 +2620,7 @@ AVCodec ff_aac_latm_decoder = {
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF,
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
.flush = flush,
};