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/*
* RTP output format |
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* Copyright (c) 2002 Fabrice Bellard |
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*
* 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
*/ |
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|
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#include "avformat.h"
#include "mpegts.h" |
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#include "internal.h" |
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#include "libavutil/random_seed.h" |
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#include "libavutil/opt.h" |
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|
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#include "rtpenc.h" |
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//#define DEBUG
|
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static const AVOption options[] = { |
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FF_RTP_FLAG_OPTS(RTPMuxContext, flags), |
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{ NULL },
};
static const AVClass rtp_muxer_class = {
.class_name = "RTP muxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
|
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#define RTCP_SR_SIZE 28
|
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static int is_supported(enum CodecID id)
{
switch(id) { |
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case CODEC_ID_H263:
case CODEC_ID_H263P: |
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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: |
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case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB: |
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case CODEC_ID_VORBIS:
case CODEC_ID_THEORA: |
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case CODEC_ID_VP8: |
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case CODEC_ID_ADPCM_G722: |
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return 1;
default:
return 0;
}
}
|
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static int rtp_write_header(AVFormatContext *s1)
{ |
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RTPMuxContext *s = s1->priv_data; |
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int max_packet_size, n; |
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AVStream *st;
if (s1->nb_streams != 1)
return -1;
st = s1->streams[0]; |
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if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
return -1;
} |
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|
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s->payload_type = ff_rtp_get_payload_type(st->codec);
if (s->payload_type < 0) |
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s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); |
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|
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s->base_timestamp = av_get_random_seed(); |
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s->timestamp = s->base_timestamp;
s->cur_timestamp = 0; |
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s->ssrc = av_get_random_seed(); |
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s->first_packet = 1; |
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s->first_rtcp_ntp_time = ff_ntp_time(); |
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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; |
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|
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max_packet_size = s1->pb->max_packet_size; |
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if (max_packet_size <= 12)
return AVERROR(EIO); |
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s->buf = av_malloc(max_packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
} |
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s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay) { |
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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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);
}
} |
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if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { |
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/* FIXME: We should round down here... */ |
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s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
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}
}
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; |
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case CODEC_ID_H264:
/* check for H.264 MP4 syntax */ |
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if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { |
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s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break; |
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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; |
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case CODEC_ID_VP8: |
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av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
"incompatible with the latest spec drafts.\n"); |
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break; |
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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; |
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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;
} |
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case CODEC_ID_AAC: |
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s->num_frames = 0; |
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default: |
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defaultcase: |
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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uint32_t rtp_ts;
|
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av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
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s->last_rtcp_ntp_time = ntp_time; |
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rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
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s1->streams[0]->time_base) + s->base_timestamp; |
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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); |
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avio_flush(s1->pb); |
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}
/* 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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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|
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av_dlog(s1, "rtp_send_data size=%d\n", len); |
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/* build the RTP header */ |
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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); |
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|
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avio_write(s1->pb, buf1, len); |
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avio_flush(s1->pb); |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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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)
{ |
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RTPMuxContext *s = s1->priv_data; |
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AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
|
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av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); |
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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) && |
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(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
rtcp_send_sr(s1, ff_ntp_time()); |
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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: |
d3d1eae6 |
rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels); |
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break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE: |
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rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels); |
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break; |
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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; |
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case CODEC_ID_MP2:
case CODEC_ID_MP3: |
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rtp_send_mpegaudio(s1, pkt->data, size); |
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break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO: |
d3d1eae6 |
ff_rtp_send_mpegvideo(s1, pkt->data, size); |
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break;
case CODEC_ID_AAC: |
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if (s->flags & FF_RTP_FLAG_MP4A_LATM) |
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ff_rtp_send_latm(s1, pkt->data, size);
else
ff_rtp_send_aac(s1, pkt->data, size); |
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break; |
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case CODEC_ID_AMR_NB:
case CODEC_ID_AMR_WB: |
d3d1eae6 |
ff_rtp_send_amr(s1, pkt->data, size); |
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break; |
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case CODEC_ID_MPEG2TS: |
d3d1eae6 |
rtp_send_mpegts_raw(s1, pkt->data, size); |
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break; |
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case CODEC_ID_H264: |
d3d1eae6 |
ff_rtp_send_h264(s1, pkt->data, size); |
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break; |
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case CODEC_ID_H263:
case CODEC_ID_H263P: |
d3d1eae6 |
ff_rtp_send_h263(s1, pkt->data, size); |
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break; |
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case CODEC_ID_VORBIS:
case CODEC_ID_THEORA:
ff_rtp_send_xiph(s1, pkt->data, size);
break; |
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case CODEC_ID_VP8:
ff_rtp_send_vp8(s1, pkt->data, size);
break; |
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default:
/* better than nothing : send the codec raw data */ |
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rtp_send_raw(s1, pkt->data, size); |
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break;
}
return 0;
}
|
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static int rtp_write_trailer(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
av_freep(&s->buf);
return 0;
}
|
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AVOutputFormat ff_rtp_muxer = { |
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"rtp", |
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NULL_IF_CONFIG_SMALL("RTP output format"), |
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NULL,
NULL, |
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sizeof(RTPMuxContext), |
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CODEC_ID_PCM_MULAW,
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet, |
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rtp_write_trailer, |
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.priv_class = &rtp_muxer_class, |
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}; |