mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-21 08:45:06 +02:00
458 lines
14 KiB
C
458 lines
14 KiB
C
/*
|
|
* 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/random_seed.h"
|
|
|
|
#include "rtpenc.h"
|
|
|
|
//#define DEBUG
|
|
|
|
#define RTCP_SR_SIZE 28
|
|
|
|
static int is_supported(enum CodecID id)
|
|
{
|
|
switch(id) {
|
|
case CODEC_ID_H263:
|
|
case CODEC_ID_H263P:
|
|
case CODEC_ID_H264:
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
case CODEC_ID_MPEG4:
|
|
case CODEC_ID_AAC:
|
|
case CODEC_ID_MP2:
|
|
case CODEC_ID_MP3:
|
|
case CODEC_ID_PCM_ALAW:
|
|
case CODEC_ID_PCM_MULAW:
|
|
case CODEC_ID_PCM_S8:
|
|
case CODEC_ID_PCM_S16BE:
|
|
case CODEC_ID_PCM_S16LE:
|
|
case CODEC_ID_PCM_U16BE:
|
|
case CODEC_ID_PCM_U16LE:
|
|
case CODEC_ID_PCM_U8:
|
|
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;
|
|
}
|
|
}
|
|
|
|
static int rtp_write_header(AVFormatContext *s1)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
int max_packet_size, n;
|
|
AVStream *st;
|
|
|
|
if (s1->nb_streams != 1)
|
|
return -1;
|
|
st = s1->streams[0];
|
|
if (!is_supported(st->codec->codec_id)) {
|
|
av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
|
|
|
|
return -1;
|
|
}
|
|
|
|
s->payload_type = ff_rtp_get_payload_type(st->codec);
|
|
if (s->payload_type < 0)
|
|
s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
|
|
|
|
s->base_timestamp = av_get_random_seed();
|
|
s->timestamp = s->base_timestamp;
|
|
s->cur_timestamp = 0;
|
|
s->ssrc = av_get_random_seed();
|
|
s->first_packet = 1;
|
|
s->first_rtcp_ntp_time = ff_ntp_time();
|
|
if (s1->start_time_realtime)
|
|
/* Round the NTP time to whole milliseconds. */
|
|
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
|
|
NTP_OFFSET_US;
|
|
|
|
max_packet_size = url_fget_max_packet_size(s1->pb);
|
|
if (max_packet_size <= 12)
|
|
return AVERROR(EIO);
|
|
s->buf = av_malloc(max_packet_size);
|
|
if (s->buf == NULL) {
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
s->max_payload_size = max_packet_size - 12;
|
|
|
|
s->max_frames_per_packet = 0;
|
|
if (s1->max_delay) {
|
|
if (st->codec->codec_type == AVMEDIA_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 == AVMEDIA_TYPE_VIDEO) {
|
|
/* FIXME: We should round down here... */
|
|
s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, 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_H264:
|
|
/* check for H.264 MP4 syntax */
|
|
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
|
|
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_WARNING, "RTP VP8 payload is still experimental\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)
|
|
s->max_frames_per_packet = 12;
|
|
if (st->codec->codec_id == 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");
|
|
return -1;
|
|
}
|
|
if (st->codec->channels != 1) {
|
|
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
|
|
return -1;
|
|
}
|
|
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);
|
|
}
|
|
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)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
uint32_t rtp_ts;
|
|
|
|
dprintf(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, RTCP_SR);
|
|
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)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
|
|
dprintf(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);
|
|
|
|
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)
|
|
{
|
|
RTPMuxContext *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);
|
|
}
|
|
}
|
|
|
|
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;
|
|
|
|
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_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
AVStream *st = s1->streams[0];
|
|
int rtcp_bytes;
|
|
int size= pkt->size;
|
|
|
|
dprintf(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;
|
|
if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
|
|
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
|
|
rtcp_send_sr(s1, ff_ntp_time());
|
|
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, pkt->data, 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, 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);
|
|
break;
|
|
case CODEC_ID_MPEG1VIDEO:
|
|
case CODEC_ID_MPEG2VIDEO:
|
|
ff_rtp_send_mpegvideo(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_AAC:
|
|
ff_rtp_send_aac(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_AMR_NB:
|
|
case CODEC_ID_AMR_WB:
|
|
ff_rtp_send_amr(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_MPEG2TS:
|
|
rtp_send_mpegts_raw(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_H264:
|
|
ff_rtp_send_h264(s1, pkt->data, size);
|
|
break;
|
|
case CODEC_ID_H263:
|
|
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);
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_write_trailer(AVFormatContext *s1)
|
|
{
|
|
RTPMuxContext *s = s1->priv_data;
|
|
|
|
av_freep(&s->buf);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat rtp_muxer = {
|
|
"rtp",
|
|
NULL_IF_CONFIG_SMALL("RTP output format"),
|
|
NULL,
|
|
NULL,
|
|
sizeof(RTPMuxContext),
|
|
CODEC_ID_PCM_MULAW,
|
|
CODEC_ID_NONE,
|
|
rtp_write_header,
|
|
rtp_write_packet,
|
|
rtp_write_trailer,
|
|
};
|