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Handle G.722 in RTP, and all the exceptions mandated in RFC 3551

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Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Martin Storsjö 2010-09-15 17:35:39 +00:00
parent 82eac2f321
commit 0048a2a8d3
4 changed files with 26 additions and 1 deletions
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2
libavformat/rtp.c
View File

@ -48,7 +48,7 @@ static const struct
{6, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
{7, "LPC", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{8, "PCMA", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
{9, "G722", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{9, "G722", AVMEDIA_TYPE_AUDIO, CODEC_ID_ADPCM_G722, 8000, 1},
{10, "L16", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
{11, "L16", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
{12, "QCELP", AVMEDIA_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},

7
libavformat/rtpdec.c
View File

@ -365,6 +365,13 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
case CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
case CODEC_ID_ADPCM_G722:
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
/* According to RFC 3551, the stream clock rate is 8000
* even if the sample rate is 16000. */
if (st->codec->sample_rate == 8000)
st->codec->sample_rate = 16000;
break;
default:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);

12
libavformat/rtpenc.c
View File

@ -56,6 +56,7 @@ static int is_supported(enum CodecID id)
case CODEC_ID_VORBIS:
case CODEC_ID_THEORA:
case CODEC_ID_VP8:
case CODEC_ID_ADPCM_G722:
return 1;
default:
return 0;
@ -148,6 +149,11 @@ static int rtp_write_header(AVFormatContext *s1)
case CODEC_ID_VP8:
av_log(s1, AV_LOG_WARNING, "RTP VP8 payload is still experimental\n");
break;
case CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
av_set_pts_info(st, 32, 1, 8000);
break;
case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
@ -382,6 +388,12 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, pkt->data, size, 2 * 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);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);

6
libavformat/sdp.c
View File

@ -419,6 +419,12 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
av_strlcatf(buff, size, "a=rtpmap:%d VP8/90000\r\n",
payload_type);
break;
case CODEC_ID_ADPCM_G722:
if (payload_type >= RTP_PT_PRIVATE)
av_strlcatf(buff, size, "a=rtpmap:%d G722/%d/%d\r\n",
payload_type,
8000, c->channels);
break;
default:
/* Nothing special to do here... */
break;