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ffmpeg.wasm-core
Commit 83a0d387 authored by Luca Abeni's avatar Luca Abeni
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Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies 

Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 9389e63c
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Showing with 357 additions and 330 deletions
libavformat/Makefile
View file @ 83a0d387
......@@ -121,9 +121,9 @@ OBJS-$(CONFIG_RM_DEMUXER) += rmdec.o
OBJS-$(CONFIG_RM_MUXER) += rmenc.o
OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o
OBJS-$(CONFIG_ROQ_MUXER) += raw.o
OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_mpv.o rtp_aac.o
OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtpenc.o rtp_mpv.o rtp_aac.o
OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtpdec.o rtp_h264.o rtp_mpv.o rtp_aac.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtpdec.o rtp_h264.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
OBJS-$(CONFIG_SHORTEN_DEMUXER) += raw.o
OBJS-$(CONFIG_SIFF_DEMUXER) += siff.o
......
libavformat/rtp.c
View file @ 83a0d387
......@@ -19,20 +19,15 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "mpegts.h"
#include "bitstream.h"
#include <unistd.h>
#include "network.h"
#include "rtp_internal.h"
#include "rtp_mpv.h"
#include "rtp_aac.h"
//#define DEBUG
#define RTCP_SR_SIZE 28
/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
AVRtpPayloadType_t AVRtpPayloadTypes[]=
{
......@@ -225,326 +220,3 @@ enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
return CODEC_ID_NONE;
}
/* rtp output */
static int rtp_write_header(AVFormatContext *s1)
{
RTPDemuxContext *s = s1->priv_data;
int payload_type, max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0];
payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
max_packet_size = url_fget_max_packet_size(s1->pb);
if (max_packet_size <= 12)
return AVERROR(EIO);
s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay) {
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if (st->codec->frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
/* FIXME: We should round down here... */
s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
}
}
av_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
break;
case CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case CODEC_ID_AAC:
s->read_buf_index = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
}
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
RTPDemuxContext *s = s1->priv_data;
uint32_t rtp_ts;
#if defined(DEBUG)
printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
s->last_rtcp_ntp_time = ntp_time;
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
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);
}
/* 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)
{
RTPDemuxContext *s = s1->priv_data;
#ifdef DEBUG
printf("rtp_send_data size=%d\n", len);
#endif
/* 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);
put_buffer(s1->pb, buf1, len);
put_flush_packet(s1->pb);
s->seq++;
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 void rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size)
{
RTPDemuxContext *s = s1->priv_data;
int len, max_packet_size, n;
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
/* not needed, but who nows */
if ((size % sample_size) != 0)
av_abort();
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 / sample_size;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
}
/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *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)
{
RTPDemuxContext *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)
{
RTPDemuxContext *s = s1->priv_data;
int len, out_len;
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;
}
}
}
/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
uint8_t *buf1= pkt->data;
#ifdef DEBUG
printf("%d: write len=%d\n", pkt->stream_index, size);
#endif
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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) &&
(av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
rtcp_send_sr(s1, av_gettime());
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
s->cur_timestamp = s->base_timestamp + pkt->pts;
switch(st->codec->codec_id) {
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
rtp_send_mpegaudio(s1, buf1, size);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
ff_rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_AAC:
ff_rtp_send_aac(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
break;
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, buf1, size);
break;
}
return 0;
}
AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
NULL,
NULL,
sizeof(RTPDemuxContext),
CODEC_ID_PCM_MULAW,
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet,
};
libavformat/rtpenc.c 0 → 100644
View file @ 83a0d387
/*
* 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 "bitstream.h"
#include <unistd.h>
#include "network.h"
#include "rtp_internal.h"
#include "rtp_mpv.h"
#include "rtp_aac.h"
//#define DEBUG
#define RTCP_SR_SIZE 28
static int rtp_write_header(AVFormatContext *s1)
{
RTPDemuxContext *s = s1->priv_data;
int payload_type, max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0];
payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
max_packet_size = url_fget_max_packet_size(s1->pb);
if (max_packet_size <= 12)
return AVERROR(EIO);
s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay) {
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if (st->codec->frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
/* FIXME: We should round down here... */
s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
}
}
av_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
break;
case CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case CODEC_ID_AAC:
s->read_buf_index = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
}
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
RTPDemuxContext *s = s1->priv_data;
uint32_t rtp_ts;
#if defined(DEBUG)
printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
s->last_rtcp_ntp_time = ntp_time;
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
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);
}
/* 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)
{
RTPDemuxContext *s = s1->priv_data;
#ifdef DEBUG
printf("rtp_send_data size=%d\n", len);
#endif
/* 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);
put_buffer(s1->pb, buf1, len);
put_flush_packet(s1->pb);
s->seq++;
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 void rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size)
{
RTPDemuxContext *s = s1->priv_data;
int len, max_packet_size, n;
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
/* not needed, but who nows */
if ((size % sample_size) != 0)
av_abort();
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 / sample_size;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
}
/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
RTPDemuxContext *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)
{
RTPDemuxContext *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)
{
RTPDemuxContext *s = s1->priv_data;
int len, out_len;
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;
}
}
}
/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
uint8_t *buf1= pkt->data;
#ifdef DEBUG
printf("%d: write len=%d\n", pkt->stream_index, size);
#endif
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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) &&
(av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
rtcp_send_sr(s1, av_gettime());
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
s->cur_timestamp = s->base_timestamp + pkt->pts;
switch(st->codec->codec_id) {
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
rtp_send_mpegaudio(s1, buf1, size);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
ff_rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_AAC:
ff_rtp_send_aac(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
break;
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, buf1, size);
break;
}
return 0;
}
AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
NULL,
NULL,
sizeof(RTPDemuxContext),
CODEC_ID_PCM_MULAW,
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet,
};
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