# HG changeset patch
# User mstorsjo
# Date 1284572139 0
# Node ID f06908662125cfb2c6fb38cb349bad45327df837
# Parent  2da74be1a3d83312d2bf7bc770269afa2baca04a
Handle G.722 in RTP, and all the exceptions mandated in RFC 3551

diff -r 2da74be1a3d8 -r f06908662125 rtp.c
--- a/rtp.c	Tue Sep 14 22:20:46 2010 +0000
+++ b/rtp.c	Wed Sep 15 17:35:39 2010 +0000
@@ -48,7 +48,7 @@
   {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 -r 2da74be1a3d8 -r f06908662125 rtpdec.c
--- a/rtpdec.c	Tue Sep 14 22:20:46 2010 +0000
+++ b/rtpdec.c	Wed Sep 15 17:35:39 2010 +0000
@@ -365,6 +365,13 @@
         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 -r 2da74be1a3d8 -r f06908662125 rtpenc.c
--- a/rtpenc.c	Tue Sep 14 22:20:46 2010 +0000
+++ b/rtpenc.c	Wed Sep 15 17:35:39 2010 +0000
@@ -56,6 +56,7 @@
     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 @@
     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 @@
     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 -r 2da74be1a3d8 -r f06908662125 sdp.c
--- a/sdp.c	Tue Sep 14 22:20:46 2010 +0000
+++ b/sdp.c	Wed Sep 15 17:35:39 2010 +0000
@@ -419,6 +419,12 @@
             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;