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

Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 12 insertions, 0 deletions

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index f5e1e3bbd3..0a2959dfcf 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);