case CODEC_ID_VORBIS:
case CODEC_ID_THEORA:
case CODEC_ID_VP8:
+ case CODEC_ID_ADPCM_G722:
return 1;
default:
return 0;
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
- max_packet_size = url_fget_max_packet_size(s1->pb);
+ max_packet_size = s1->pb->max_packet_size;
if (max_packet_size <= 12)
return AVERROR(EIO);
s->buf = av_malloc(max_packet_size);
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:
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
- dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
+ av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
s->last_rtcp_ntp_time = 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_be16(s1->pb, 6); /* length in words - 1 */
- put_be32(s1->pb, s->ssrc);
- put_be32(s1->pb, ntp_time / 1000000);
- put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
- put_be32(s1->pb, rtp_ts);
- put_be32(s1->pb, s->packet_count);
- put_be32(s1->pb, s->octet_count);
- put_flush_packet(s1->pb);
+ avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, RTCP_SR);
+ avio_wb16(s1->pb, 6); /* length in words - 1 */
+ avio_wb32(s1->pb, s->ssrc);
+ avio_wb32(s1->pb, ntp_time / 1000000);
+ avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
+ avio_wb32(s1->pb, rtp_ts);
+ avio_wb32(s1->pb, s->packet_count);
+ avio_wb32(s1->pb, s->octet_count);
+ avio_flush(s1->pb);
}
/* send an rtp packet. sequence number is incremented, but the caller
{
RTPMuxContext *s = s1->priv_data;
- dprintf(s1, "rtp_send_data size=%d\n", len);
+ av_dlog(s1, "rtp_send_data size=%d\n", len);
/* build the RTP header */
- put_byte(s1->pb, (RTP_VERSION << 6));
- put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
- put_be16(s1->pb, s->seq);
- put_be32(s1->pb, s->timestamp);
- put_be32(s1->pb, s->ssrc);
+ avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
+ avio_wb16(s1->pb, s->seq);
+ avio_wb32(s1->pb, s->timestamp);
+ avio_wb32(s1->pb, s->ssrc);
- put_buffer(s1->pb, buf1, len);
- put_flush_packet(s1->pb);
+ avio_write(s1->pb, buf1, len);
+ avio_flush(s1->pb);
s->seq++;
s->octet_count += len;
int rtcp_bytes;
int size= pkt->size;
- dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
+ av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
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);
return 0;
}
-AVOutputFormat rtp_muxer = {
+AVOutputFormat ff_rtp_muxer = {
"rtp",
NULL_IF_CONFIG_SMALL("RTP output format"),
NULL,