Merge remote-tracking branch 'qatar/master'

* qatar/master: (24 commits)
  flvdec: remove incomplete, disabled seeking code
  mem: add support for _aligned_malloc() as found on Windows
  lavc: Extend the documentation for avcodec_init_packet
  flvdec: remove incomplete, disabled seeking code
  http: replace atoll() with strtoll()
  mpegts: remove unused/incomplete/broken seeking code
  af_amix: allow float planar sample format as input
  af_amix: use AVFloatDSPContext.vector_fmac_scalar()
  float_dsp: add x86-optimized functions for vector_fmac_scalar()
  float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
  lavr: Add x86-optimized function for flt to s32 conversion
  lavr: Add x86-optimized function for flt to s16 conversion
  lavr: Add x86-optimized functions for s32 to flt conversion
  lavr: Add x86-optimized functions for s32 to s16 conversion
  lavr: Add x86-optimized functions for s16 to flt conversion
  lavr: Add x86-optimized function for s16 to s32 conversion
  rtpenc: Support packetizing iLBC
  rtpdec: Add a depacketizer for iLBC
  Implement the iLBC storage file format
  mov: Support muxing/demuxing iLBC
  ...

Conflicts:
	Changelog
	configure
	libavcodec/avcodec.h
	libavcodec/dsputil.c
	libavcodec/version.h
	libavformat/movenc.c
	libavformat/mpegts.c
	libavformat/version.h
	libavutil/mem.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-06-19 20:52:00 +02:00
44 changed files with 952 additions and 231 deletions

View File

@ -74,6 +74,7 @@ static int is_supported(enum CodecID id)
case CODEC_ID_VP8:
case CODEC_ID_ADPCM_G722:
case CODEC_ID_ADPCM_G726:
case CODEC_ID_ILBC:
return 1;
default:
return 0;
@ -187,6 +188,16 @@ static int rtp_write_header(AVFormatContext *s1)
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 1;
s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
s->max_payload_size / st->codec->block_align);
goto defaultcase;
case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
@ -395,6 +406,36 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
}
}
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int frame_duration = av_get_audio_frame_duration(st->codec, 0);
int frame_size = st->codec->block_align;
int frames = size / frame_size;
while (frames > 0) {
int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
if (!s->num_frames) {
s->buf_ptr = s->buf;
s->timestamp = s->cur_timestamp;
}
memcpy(s->buf_ptr, buf, n * frame_size);
frames -= n;
s->num_frames += n;
s->buf_ptr += n * frame_size;
buf += n * frame_size;
s->cur_timestamp += n * frame_duration;
if (s->num_frames == s->max_frames_per_packet) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
s->num_frames = 0;
}
}
return 0;
}
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPMuxContext *s = s1->priv_data;
@ -483,6 +524,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case CODEC_ID_VP8:
ff_rtp_send_vp8(s1, pkt->data, size);
break;
case CODEC_ID_ILBC:
rtp_send_ilbc(s1, pkt->data, size);
break;
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);