0,0 → 1,608 |
/* |
* 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_H263: |
case AV_CODEC_ID_H263P: |
case AV_CODEC_ID_H264: |
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_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_ADPCM_G722: |
case AV_CODEC_ID_ADPCM_G726: |
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; |
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->codec->codec_id)) { |
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->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->codec, -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) |
/* 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 (st->codec->flags & CODEC_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 %d too low\n", s1->packet_size); |
return AVERROR(EIO); |
} |
s->buf = av_malloc(s1->packet_size); |
if (s->buf == NULL) { |
return AVERROR(ENOMEM); |
} |
s->max_payload_size = s1->packet_size - 12; |
|
s->max_frames_per_packet = 0; |
if (s1->max_delay > 0) { |
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
int frame_size = av_get_audio_frame_duration(st->codec, 0); |
if (!frame_size) |
frame_size = st->codec->frame_size; |
if (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_q_rnd(s1->max_delay, |
AV_TIME_BASE_Q, |
(AVRational){ frame_size, st->codec->sample_rate }, |
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); |
} |
} |
|
avpriv_set_pts_info(st, 32, 1, 90000); |
switch(st->codec->codec_id) { |
case AV_CODEC_ID_MP2: |
case AV_CODEC_ID_MP3: |
s->buf_ptr = s->buf + 4; |
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; |
s->buf_ptr = s->buf; |
break; |
case AV_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 AV_CODEC_ID_VORBIS: |
case AV_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 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->codec->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->codec->block_align != 38 && st->codec->block_align != 50) { |
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n"); |
goto fail; |
} |
if (!s->max_frames_per_packet) |
s->max_frames_per_packet = 1; |
s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, |
s->max_payload_size / st->codec->block_align); |
goto defaultcase; |
case AV_CODEC_ID_AMR_NB: |
case AV_CODEC_ID_AMR_WB: |
if (!s->max_frames_per_packet) |
s->max_frames_per_packet = 12; |
if (st->codec->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->codec->channels != 1) { |
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); |
goto fail; |
} |
case AV_CODEC_ID_AAC: |
s->num_frames = 0; |
default: |
defaultcase: |
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); |
} |
s->buf_ptr = s->buf; |
break; |
} |
|
return 0; |
|
fail: |
av_freep(&s->buf); |
return AVERROR(EINVAL); |
} |
|
/* 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; |
|
av_dlog(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; |
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_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); |
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); |
} |
|
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_dlog(s1, "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; |
|
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_duration(st->codec, 0); |
int frame_size = st->codec->block_align; |
int frames = size / frame_size; |
|
while (frames > 0) { |
int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); |
|
if (!s->num_frames) { |
s->buf_ptr = s->buf; |
s->timestamp = s->cur_timestamp; |
} |
memcpy(s->buf_ptr, buf, n * frame_size); |
frames -= n; |
s->num_frames += n; |
s->buf_ptr += n * frame_size; |
buf += n * frame_size; |
s->cur_timestamp += n * 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_dlog(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))) && |
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) { |
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 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->codec->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->codec->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->codec->channels); |
case AV_CODEC_ID_ADPCM_G726: |
return rtp_send_samples(s1, pkt->data, size, |
st->codec->bits_per_coded_sample * st->codec->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_H264: |
ff_rtp_send_h264(s1, pkt->data, size); |
break; |
case AV_CODEC_ID_H263: |
if (s->flags & FF_RTP_FLAG_RFC2190) { |
int mb_info_size = 0; |
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_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_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; |
|
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, |
}; |