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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  mpegvideo: reduce excessive inlining of mpeg_motion()
  mpegvideo: convert mpegvideo_common.h to a .c file
  build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
  Move MASK_ABS macro to libavcodec/mathops.h
  x86: move MANGLE() and related macros to libavutil/x86/asm.h
  x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
  aacdec: Don't fall back to the old output configuration when no old configuration is present.
  rtmp: Add message tracking
  rtsp: Support mpegts in raw udp packets
  rtsp: Support receiving plain data over UDP without any RTP encapsulation
  rtpdec: Remove an unused include
  rtpenc: Remove an av_abort() that depends on user-supplied data
  vsrc_movie: discourage its use with avconv.
  avconv: allow no input files.
  avconv: prevent invalid reads in transcode_init()
  avconv: rename OutputStream.is_past_recording_time to finished.

Conflicts:
	configure
	doc/filters.texi
	ffmpeg.c
	ffmpeg.h
	libavcodec/Makefile
	libavcodec/aacdec.c
	libavcodec/mpegvideo.c
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-08-09 19:09:39 +02:00
parent e1a983e601 f69f4036f8
commit 9f088a1ed4
56 changed files with 533 additions and 445 deletions
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21
libavformat/rtpenc.c
View File

@ -281,8 +281,8 @@ 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_bits)
static int rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size_bits)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
@ -292,7 +292,7 @@ static void rtp_send_samples(AVFormatContext *s1,
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();
return AVERROR(EINVAL);
n = 0;
while (size > 0) {
s->buf_ptr = s->buf;
@ -307,6 +307,7 @@ static void rtp_send_samples(AVFormatContext *s1,
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
return 0;
}
static void rtp_send_mpegaudio(AVFormatContext *s1,
@ -461,25 +462,21 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_S8:
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
case AV_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_bits is 8 bits per stream
* clock. */
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_ADPCM_G726:
rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);