asfdec: ensure that the whole tag is read.

[ffmpeg] / libavformat / rtpenc.c

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index a6ba3e50b11658bcf7d75840b70ae88fab092681..839a05c338266456f137ff0b2c79a13ce1c9a0ad 100644 (file)
--- 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;
@@ -146,7 +147,13 @@ static int rtp_write_header(AVFormatContext *s1)
         s->num_frames = 0;
         goto defaultcase;
     case CODEC_ID_VP8:
-        av_log(s1, AV_LOG_WARNING, "RTP VP8 payload is still experimental\n");
+        av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
+                                 "incompatible with the latest spec drafts.\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:
@@ -191,7 +198,7 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
                           s1->streams[0]->time_base) + s->base_timestamp;
     put_byte(s1->pb, (RTP_VERSION << 6));
-    put_byte(s1->pb, 200);
+    put_byte(s1->pb, RTCP_SR);
     put_be16(s1->pb, 6); /* length in words - 1 */
     put_be32(s1->pb, s->ssrc);
     put_be32(s1->pb, ntp_time / 1000000);
@@ -382,6 +389,12 @@ 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);
@@ -431,7 +444,7 @@ static int rtp_write_trailer(AVFormatContext *s1)
     return 0;
 }
 
-AVOutputFormat rtp_muxer = {
+AVOutputFormat ff_rtp_muxer = {
     "rtp",
     NULL_IF_CONFIG_SMALL("RTP output format"),
     NULL,