diff options
-rw-r--r-- | libavformat/rtp.c | 2 | ||||
-rw-r--r-- | libavformat/rtpdec.c | 7 | ||||
-rw-r--r-- | libavformat/rtpenc.c | 12 | ||||
-rw-r--r-- | libavformat/sdp.c | 6 |
4 files changed, 26 insertions, 1 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index a8dcfd7..70c5e99 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -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}, diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index debc14c..942b8d7 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -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); diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index f5e1e3b..0a2959d 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -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); diff --git a/libavformat/sdp.c b/libavformat/sdp.c index f7c1193..a4bf7fb 100644 --- a/libavformat/sdp.c +++ b/libavformat/sdp.c @@ -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; |