* RTP output format
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "internal.h"
#include "libavutil/random_seed.h"
-#include <unistd.h>
-
#include "rtpenc.h"
//#define DEBUG
case CODEC_ID_MPEG2TS:
case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB:
+ 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->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
+ s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
+ s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
+ s->num_frames = 0;
+ goto defaultcase;
+ case CODEC_ID_VP8:
+ 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:
if (!s->max_frames_per_packet)
case CODEC_ID_AAC:
s->num_frames = 0;
default:
+defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
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);
case CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size);
break;
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ ff_rtp_send_xiph(s1, pkt->data, size);
+ break;
+ case CODEC_ID_VP8:
+ ff_rtp_send_vp8(s1, pkt->data, size);
+ break;
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);
return 0;
}
-AVOutputFormat rtp_muxer = {
+AVOutputFormat ff_rtp_muxer = {
"rtp",
NULL_IF_CONFIG_SMALL("RTP output format"),
NULL,
projects / ffmpeg / blobdiff