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FFmpeg/libavformat/rtpenc.c

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/*
* RTP output format
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "mpegts.h"
#include "internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/random_seed.h"
#include "libavutil/opt.h"
#include "rtpenc.h"
static const AVOption options[] = {
FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
{ "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
{ "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
{ "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
{ NULL },
};
static const AVClass rtp_muxer_class = {
.class_name = "RTP muxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
#define RTCP_SR_SIZE 28
static int is_supported(enum AVCodecID id)
{
switch(id) {
case AV_CODEC_ID_DIRAC:
case AV_CODEC_ID_H261:
case AV_CODEC_ID_H263:
case AV_CODEC_ID_H263P:
case AV_CODEC_ID_H264:
case AV_CODEC_ID_HEVC:
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
case AV_CODEC_ID_MPEG4:
case AV_CODEC_ID_AAC:
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_MULAW:
case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_S24BE:
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_MPEG2TS:
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
case AV_CODEC_ID_VP8:
case AV_CODEC_ID_VP9:
case AV_CODEC_ID_ADPCM_G722:
case AV_CODEC_ID_ADPCM_G726:
case AV_CODEC_ID_ADPCM_G726LE:
case AV_CODEC_ID_ILBC:
case AV_CODEC_ID_MJPEG:
case AV_CODEC_ID_SPEEX:
case AV_CODEC_ID_OPUS:
return 1;
default:
return 0;
}
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n, ret = AVERROR(EINVAL);
AVStream *st;
if (s1->nb_streams != 1) {
av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
if (!is_supported(st->codecpar->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
return -1;
}
if (s->payload_type < 0) {
/* Re-validate non-dynamic payload types */
if (st->id < RTP_PT_PRIVATE)
st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
s->payload_type = st->id;
} else {
/* private option takes priority */
st->id = s->payload_type;
}
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
// Pick a random sequence start number, but in the lower end of the
// available range, so that any wraparound doesn't happen immediately.
// (Immediate wraparound would be an issue for SRTP.)
if (s->seq < 0) {
if (s1->flags & AVFMT_FLAG_BITEXACT) {
s->seq = 0;
} else
s->seq = av_get_random_seed() & 0x0fff;
} else
s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
if (s1->packet_size) {
if (s1->pb->max_packet_size)
s1->packet_size = FFMIN(s1->packet_size,
s1->pb->max_packet_size);
} else
s1->packet_size = s1->pb->max_packet_size;
if (s1->packet_size <= 12) {
av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
if (!s->buf) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
} else {
avpriv_set_pts_info(st, 32, 1, 90000);
}
s->buf_ptr = s->buf;
switch(st->codecpar->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
avpriv_set_pts_info(st, 32, 1, 90000);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
break;
case AV_CODEC_ID_DIRAC:
if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
av_log(s, AV_LOG_ERROR,
"Packetizing VC-2 is experimental and does not use all values "
"of the specification "
"(even though most receivers may handle it just fine). "
"Please set -strict experimental in order to enable it.\n");
ret = AVERROR_EXPERIMENTAL;
goto fail;
}
break;
case AV_CODEC_ID_H261:
if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
av_log(s, AV_LOG_ERROR,
"Packetizing H.261 is experimental and produces incorrect "
"packetization for cases where GOBs don't fit into packets "
"(even though most receivers may handle it just fine). "
"Please set -f_strict experimental in order to enable it.\n");
ret = AVERROR_EXPERIMENTAL;
goto fail;
}
break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
}
break;
case AV_CODEC_ID_HEVC:
/* Only check for the standardized hvcC version of extradata, keeping
* things simple and similar to the avcC/H.264 case above, instead
* of trying to handle the pre-standardization versions (as in
* libavcodec/hevc.c). */
if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VP9:
if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
av_log(s, AV_LOG_ERROR,
"Packetizing VP9 is experimental and its specification is "
"still in draft state. "
"Please set -strict experimental in order to enable it.\n");
ret = AVERROR_EXPERIMENTAL;
goto fail;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
s->max_frames_per_packet = 15;
break;
case AV_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. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_OPUS:
if (st->codecpar->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_CODEC_ID_ILBC:
if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
break;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
s->max_frames_per_packet = 50;
if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
if (st->codecpar->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
break;
case AV_CODEC_ID_AAC:
s->max_frames_per_packet = 50;
break;
default:
break;
}
return 0;
fail:
av_freep(&s->buf);
return ret;
}
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
{
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\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;
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);
if (s->cname) {
int len = FFMIN(strlen(s->cname), 255);
avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
avio_w8(s1->pb, RTCP_SDES);
avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
avio_wb32(s1->pb, s->ssrc);
avio_w8(s1->pb, 0x01); /* CNAME */
avio_w8(s1->pb, len);
avio_write(s1->pb, s->cname, len);
avio_w8(s1->pb, 0); /* END */
for (len = (7 + len) % 4; len % 4; len++)
avio_w8(s1->pb, 0);
}
if (bye) {
avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
avio_w8(s1->pb, RTCP_BYE);
avio_wb16(s1->pb, 1); /* length in words - 1 */
avio_wb32(s1->pb, s->ssrc);
}
avio_flush(s1->pb);
}
/* send an rtp packet. sequence number is incremented, but the caller
must update the timestamp itself */
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPMuxContext *s = s1->priv_data;
av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
/* build the RTP header */
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);
avio_write(s1->pb, buf1, len);
avio_flush(s1->pb);
s->seq = (s->seq + 1) & 0xffff;
s->octet_count += len;
s->packet_count++;
}
/* send an integer number of samples and compute time stamp and fill
the rtp send buffer before sending. */
static int rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size_bits)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
/* Calculate the number of bytes to get samples aligned on a byte border */
int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
return AVERROR(EINVAL);
n = 0;
while (size > 0) {
s->buf_ptr = s->buf;
len = FFMIN(max_packet_size, size);
/* copy data */
memcpy(s->buf_ptr, buf1, len);
s->buf_ptr += len;
buf1 += len;
size -= len;
s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
return 0;
}
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
int len, count, max_packet_size;
max_packet_size = s->max_payload_size;
/* test if we must flush because not enough space */
len = (s->buf_ptr - s->buf);
if ((len + size) > max_packet_size) {
if (len > 4) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->buf_ptr = s->buf + 4;
}
}
if (s->buf_ptr == s->buf + 4) {
s->timestamp = s->cur_timestamp;
}
/* add the packet */
if (size > max_packet_size) {
/* big packet: fragment */
count = 0;
while (size > 0) {
len = max_packet_size - 4;
if (len > size)
len = size;
/* build fragmented packet */
s->buf[0] = 0;
s->buf[1] = 0;
s->buf[2] = count >> 8;
s->buf[3] = count;
memcpy(s->buf + 4, buf1, len);
ff_rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
}
} else {
if (s->buf_ptr == s->buf + 4) {
/* no fragmentation possible */
s->buf[0] = 0;
s->buf[1] = 0;
s->buf[2] = 0;
s->buf[3] = 0;
}
memcpy(s->buf_ptr, buf1, size);
s->buf_ptr += size;
}
}
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size;
max_packet_size = s->max_payload_size;
while (size > 0) {
len = max_packet_size;
if (len > size)
len = size;
s->timestamp = s->cur_timestamp;
ff_rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
}
}
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPMuxContext *s = s1->priv_data;
int len, out_len;
s->timestamp = s->cur_timestamp;
while (size >= TS_PACKET_SIZE) {
len = s->max_payload_size - (s->buf_ptr - s->buf);
if (len > size)
len = size;
memcpy(s->buf_ptr, buf1, len);
buf1 += len;
size -= len;
s->buf_ptr += len;
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
ff_rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
}
static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
int frame_size = st->codecpar->block_align;
int frames = size / frame_size;
while (frames > 0) {
if (s->num_frames > 0 &&
av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
s1->max_delay, AV_TIME_BASE_Q) >= 0) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
s->num_frames = 0;
}
if (!s->num_frames) {
s->buf_ptr = s->buf;
s->timestamp = s->cur_timestamp;
}
memcpy(s->buf_ptr, buf, frame_size);
frames--;
s->num_frames++;
s->buf_ptr += frame_size;
buf += frame_size;
s->cur_timestamp += frame_duration;
if (s->num_frames == s->max_frames_per_packet) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
s->num_frames = 0;
}
}
return 0;
}
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
av_log(s1, AV_LOG_TRACE, "%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;
if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
rtcp_send_sr(s1, ff_ntp_time(), 0);
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
s->cur_timestamp = s->base_timestamp + pkt->pts;
switch(st->codecpar->codec_id) {
case AV_CODEC_ID_PCM_MULAW:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_S8:
return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
case AV_CODEC_ID_PCM_S24BE:
return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels);
case AV_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_bits is 8 bits per stream
* clock. */
return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
case AV_CODEC_ID_ADPCM_G726:
case AV_CODEC_ID_ADPCM_G726LE:
return rtp_send_samples(s1, pkt->data, size,
st->codecpar->bits_per_coded_sample * st->codecpar->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
ff_rtp_send_mpegvideo(s1, pkt->data, size);
break;
case AV_CODEC_ID_AAC:
if (s->flags & FF_RTP_FLAG_MP4A_LATM)
ff_rtp_send_latm(s1, pkt->data, size);
else
ff_rtp_send_aac(s1, pkt->data, size);
break;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
ff_rtp_send_amr(s1, pkt->data, size);
break;
case AV_CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, pkt->data, size);
break;
case AV_CODEC_ID_DIRAC:
ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
break;
case AV_CODEC_ID_H264:
ff_rtp_send_h264_hevc(s1, pkt->data, size);
break;
case AV_CODEC_ID_H261:
ff_rtp_send_h261(s1, pkt->data, size);
break;
case AV_CODEC_ID_H263:
if (s->flags & FF_RTP_FLAG_RFC2190) {
int mb_info_size;
const uint8_t *mb_info =
av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
&mb_info_size);
ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
break;
}
/* Fallthrough */
case AV_CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size);
break;
case AV_CODEC_ID_HEVC:
ff_rtp_send_h264_hevc(s1, pkt->data, size);
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
ff_rtp_send_xiph(s1, pkt->data, size);
break;
case AV_CODEC_ID_VP8:
ff_rtp_send_vp8(s1, pkt->data, size);
break;
case AV_CODEC_ID_VP9:
ff_rtp_send_vp9(s1, pkt->data, size);
break;
case AV_CODEC_ID_ILBC:
rtp_send_ilbc(s1, pkt->data, size);
break;
case AV_CODEC_ID_MJPEG:
ff_rtp_send_jpeg(s1, pkt->data, size);
break;
case AV_CODEC_ID_OPUS:
if (size > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR,
"Packet size %d too large for max RTP payload size %d\n",
size, s->max_payload_size);
return AVERROR(EINVAL);
}
/* Intentional fallthrough */
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);
break;
}
return 0;
}
static int rtp_write_trailer(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
/* If the caller closes and recreates ->pb, this might actually
* be NULL here even if it was successfully allocated at the start. */
if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
rtcp_send_sr(s1, ff_ntp_time(), 1);
av_freep(&s->buf);
return 0;
}
AVOutputFormat ff_rtp_muxer = {
.name = "rtp",
.long_name = NULL_IF_CONFIG_SMALL("RTP output"),
.priv_data_size = sizeof(RTPMuxContext),
.audio_codec = AV_CODEC_ID_PCM_MULAW,
.video_codec = AV_CODEC_ID_MPEG4,
.write_header = rtp_write_header,
.write_packet = rtp_write_packet,
.write_trailer = rtp_write_trailer,
.priv_class = &rtp_muxer_class,
.flags = AVFMT_TS_NONSTRICT,
};