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third_party_ffmpeg/libavformat/rtpenc.c
Martin Storsjö 35b0169484 RTP/VP8: Update the warning about the VP8 payload 
The current implementation is incompatible with the latest spec drafts,
this should be communicated clearly to the user.

Originally committed as revision 25887 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 11:45:57 +00:00

459 lines
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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/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_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)
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,
};
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