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Merge remote-tracking branch 'qatar/master'

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* qatar/master: (25 commits)
  rtpenc: Add support for G726 audio
  rtpdec: Interpret the different G726 names as bits_per_coded_sample
  rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
  rtpenc: Cast a rescaling parameter to int64_t
  h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
  ARM: fix indentation in ff_dsputil_init_neon()
  ARM: NEON put/avg_pixels8/16 cosmetics
  ARM: add remaining NEON avg_pixels8/16 functions
  ARM: clean up NEON put/avg_pixels macros
  fate: split acodec-pcm into individual tests
  swscale: #include "libavutil/mathematics.h"
  pmpdec: don't use deprecated av_set_pts_info.
  rv34: align temporary block of "dct" coefs
  Add PlayStation Portable PMP format demuxer
  proto: Realign struct initializers
  proto: Use .priv_data_size to allocate the private context
  mmsh: Properly clean up if the second ffurl_alloc failed
  rtmp: Clean up properly if the handshake failed
  md5proto: Remove the get_file_handle function
  applehttpproto: Use the close function if the open function fails
  ...

Conflicts:
	libavcodec/vble.c
	libavformat/mmsh.c
	libavformat/pmpdec.c
	libavformat/udp.c
	tests/ref/acodec/pcm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-12-02 00:51:11 +01:00
parent 8b08f81949 04403ec2e4
commit 7b0b10ce41
42 changed files with 462 additions and 487 deletions
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28
libavformat/rtpenc.c
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@@ -72,6 +72,7 @@ static int is_supported(enum CodecID id)
case CODEC_ID_THEORA:
case CODEC_ID_VP8:
case CODEC_ID_ADPCM_G722:
case CODEC_ID_ADPCM_G726:
return 1;
default:
return 0;
@@ -121,7 +122,7 @@ static int rtp_write_header(AVFormatContext *s1)
if (st->codec->frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
@@ -248,14 +249,16 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
/* send an integer number of samples and compute time stamp and fill
the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size)
const uint8_t *buf1, int size, int sample_size_bits)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
/* Calculate the number of bytes to get samples aligned on a byte border */
int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
/* not needed, but who nows */
if ((size % sample_size) != 0)
max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
av_abort();
n = 0;
while (size > 0) {
@@ -267,7 +270,7 @@ static void rtp_send_samples(AVFormatContext *s1,
s->buf_ptr += len;
buf1 += len;
size -= len;
s->timestamp = s->cur_timestamp + n / sample_size;
s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
@@ -394,19 +397,24 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
break;
case CODEC_ID_ADPCM_G722:
/* The actual sample size is half a byte per sample, but since the
* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
* the correct parameter for send_samples is 1 byte per stream clock. */
rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
* the correct parameter for send_samples_bits is 8 bits per stream
* clock. */
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
case CODEC_ID_ADPCM_G726:
rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3: