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srs/trunk/src/app/srs_app_mpegts_udp.cpp

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/**
* The MIT License (MIT)
*
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* Copyright (c) 2013-2017 OSSRS(winlin)
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*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
* the Software, and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <srs_app_mpegts_udp.hpp>
#ifdef SRS_AUTO_STREAM_CASTER
#include <stdlib.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
using namespace std;
#include <srs_app_config.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_log.hpp>
#include <srs_app_config.hpp>
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#include <srs_kernel_ts.hpp>
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#include <srs_kernel_buffer.hpp>
#include <srs_kernel_ts.hpp>
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#include <srs_kernel_stream.hpp>
#include <srs_kernel_file.hpp>
#include <srs_core_autofree.hpp>
#include <srs_kernel_utility.hpp>
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#include <srs_rtmp_stack.hpp>
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#include <srs_app_st.hpp>
#include <srs_protocol_utility.hpp>
#include <srs_app_utility.hpp>
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#include <srs_protocol_amf0.hpp>
#include <srs_raw_avc.hpp>
#include <srs_app_pithy_print.hpp>
#include <srs_app_rtmp_conn.hpp>
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#include <srs_protocol_utility.hpp>
SrsMpegtsQueue::SrsMpegtsQueue()
{
nb_audios = nb_videos = 0;
}
SrsMpegtsQueue::~SrsMpegtsQueue()
{
std::map<int64_t, SrsSharedPtrMessage*>::iterator it;
for (it = msgs.begin(); it != msgs.end(); ++it) {
SrsSharedPtrMessage* msg = it->second;
srs_freep(msg);
}
msgs.clear();
}
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srs_error_t SrsMpegtsQueue::push(SrsSharedPtrMessage* msg)
{
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srs_error_t err = srs_success;
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// TODO: FIXME: use right way.
for (int i = 0; i < 10; i++) {
if (msgs.find(msg->timestamp) == msgs.end()) {
break;
}
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// adjust the ts, add 1ms.
msg->timestamp += 1;
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if (i >= 5) {
srs_warn("mpegts: free the msg for dts exists, dts=%" PRId64, msg->timestamp);
srs_freep(msg);
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return err;
}
}
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if (msg->is_audio()) {
nb_audios++;
}
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if (msg->is_video()) {
nb_videos++;
}
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msgs[msg->timestamp] = msg;
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return err;
}
SrsSharedPtrMessage* SrsMpegtsQueue::dequeue()
{
// got 2+ videos and audios, ok to dequeue.
bool av_ok = nb_videos >= 2 && nb_audios >= 2;
// 100 videos about 30s, while 300 audios about 30s
bool av_overflow = nb_videos > 100 || nb_audios > 300;
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if (av_ok || av_overflow) {
std::map<int64_t, SrsSharedPtrMessage*>::iterator it = msgs.begin();
SrsSharedPtrMessage* msg = it->second;
msgs.erase(it);
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if (msg->is_audio()) {
nb_audios--;
}
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if (msg->is_video()) {
nb_videos--;
}
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return msg;
}
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return NULL;
}
SrsMpegtsOverUdp::SrsMpegtsOverUdp(SrsConfDirective* c)
{
context = new SrsTsContext();
buffer = new SrsSimpleStream();
output = _srs_config->get_stream_caster_output(c);
sdk = NULL;
avc = new SrsRawH264Stream();
aac = new SrsRawAacStream();
h264_sps_changed = false;
h264_pps_changed = false;
h264_sps_pps_sent = false;
queue = new SrsMpegtsQueue();
pprint = SrsPithyPrint::create_caster();
}
SrsMpegtsOverUdp::~SrsMpegtsOverUdp()
{
close();
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srs_freep(buffer);
srs_freep(context);
srs_freep(avc);
srs_freep(aac);
srs_freep(queue);
srs_freep(pprint);
}
srs_error_t SrsMpegtsOverUdp::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
{
std::string peer_ip = inet_ntoa(from->sin_addr);
int peer_port = ntohs(from->sin_port);
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// append to buffer.
buffer->append(buf, nb_buf);
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srs_error_t err = on_udp_bytes(peer_ip, peer_port, buf, nb_buf);
if (err != srs_success) {
return srs_error_wrap(err, "process udp");
}
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return err;
}
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srs_error_t SrsMpegtsOverUdp::on_udp_bytes(string host, int port, char* buf, int nb_buf)
{
srs_error_t err = srs_success;
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// collect nMB data to parse in a time.
// TODO: FIXME: comment the following for release.
//if (buffer->length() < 3 * 1024 * 1024) return ret;
// TODO: FIXME: remove the debug to file.
#if 0
SrsFileWriter fw;
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if ((err = fw.open("latest.ts")) != srs_success) {
return srs_error_wrap(err, "open file");
}
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if ((err = fw.write(buffer->bytes(), buffer->length(), NULL)) != srs_success) {
return srs_error_wrap(err, "write data");
}
fw.close();
#endif
#if 0
SrsFileReader fr;
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if ((err = fr.open("latest.ts")) != srs_success) {
return srs_error_wrap(err, "open file");
}
buffer->erase(buffer->length());
int nb_fbuf = fr.filesize();
char* fbuf = new char[nb_fbuf];
SrsAutoFreeA(char, fbuf);
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if ((err = fr.read(fbuf, nb_fbuf, NULL)) != srs_success) {
return srs_error_wrap(err, "read data");
}
fr.close();
buffer->append(fbuf, nb_fbuf);
#endif
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// find the sync byte of mpegts.
char* p = buffer->bytes();
for (int i = 0; i < buffer->length(); i++) {
if (p[i] != 0x47) {
continue;
}
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if (i > 0) {
buffer->erase(i);
}
break;
}
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// drop ts packet when size not modulus by 188
if (buffer->length() < SRS_TS_PACKET_SIZE) {
srs_warn("udp: wait %s:%d packet %d/%d bytes", host.c_str(), port, nb_buf, buffer->length());
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return err;
}
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// use stream to parse ts packet.
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int nb_packet = buffer->length() / SRS_TS_PACKET_SIZE;
for (int i = 0; i < nb_packet; i++) {
char* p = buffer->bytes() + (i * SRS_TS_PACKET_SIZE);
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SrsBuffer* stream = new SrsBuffer(p, SRS_TS_PACKET_SIZE);
SrsAutoFree(SrsBuffer, stream);
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// process each ts packet
if ((err = context->decode(stream, this)) != srs_success) {
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srs_warn("parse ts packet err=%s", srs_error_desc(err).c_str());
srs_error_reset(err);
continue;
}
}
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// erase consumed bytes
if (nb_packet > 0) {
buffer->erase(nb_packet * SRS_TS_PACKET_SIZE);
}
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return err;
}
srs_error_t SrsMpegtsOverUdp::on_ts_message(SrsTsMessage* msg)
{
srs_error_t err = srs_success;
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pprint->elapse();
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// about the bytes of msg, specified by elementary stream which indicates by PES_packet_data_byte and stream_id
// for example, when SrsTsStream of SrsTsChannel indicates stream_type is SrsTsStreamVideoMpeg4 and SrsTsStreamAudioMpeg4,
// the elementary stream can be mux in "2.11 Carriage of ISO/IEC 14496 data" in hls-mpeg-ts-iso13818-1.pdf, page 103
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// @remark, the most popular stream_id is 0xe0 for h.264 over mpegts, which indicates the stream_id is video and
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// stream_number is 0, where I guess the elementary is specified in annexb format(ISO_IEC_14496-10-AVC-2003.pdf, page 211).
// because when audio stream_number is 0, the elementary is ADTS(ISO_IEC_14496-3-AAC-2001.pdf, page 75, 1.A.2.2 ADTS).
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// about the bytes of PES_packet_data_byte, defined in hls-mpeg-ts-iso13818-1.pdf, page 58
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// PES_packet_data_byte "C PES_packet_data_bytes shall be contiguous bytes of data from the elementary stream
// indicated by the packets stream_id or PID. When the elementary stream data conforms to ITU-T
// Rec. H.262 | ISO/IEC 13818-2 or ISO/IEC 13818-3, the PES_packet_data_bytes shall be byte aligned to the bytes of this
// Recommendation | International Standard. The byte-order of the elementary stream shall be preserved. The number of
// PES_packet_data_bytes, N, is specified by the PES_packet_length field. N shall be equal to the value indicated in the
// PES_packet_length minus the number of bytes between the last byte of the PES_packet_length field and the first
// PES_packet_data_byte.
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//
// In the case of a private_stream_1, private_stream_2, ECM_stream, or EMM_stream, the contents of the
// PES_packet_data_byte field are user definable and will not be specified by ITU-T | ISO/IEC in the future.
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// about the bytes of stream_id, define in hls-mpeg-ts-iso13818-1.pdf, page 49
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// stream_id "C In Program Streams, the stream_id specifies the type and number of the elementary stream as defined by the
// stream_id Table 2-18. In Transport Streams, the stream_id may be set to any valid value which correctly describes the
// elementary stream type as defined in Table 2-18. In Transport Streams, the elementary stream type is specified in the
// Program Specific Information as specified in 2.4.4.
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// about the stream_id table, define in Table 2-18 "C Stream_id assignments, hls-mpeg-ts-iso13818-1.pdf, page 52.
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//
// 110x xxxx
// ISO/IEC 13818-3 or ISO/IEC 11172-3 or ISO/IEC 13818-7 or ISO/IEC
// 14496-3 audio stream number x xxxx
// ((sid >> 5) & 0x07) == SrsTsPESStreamIdAudio
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//
// 1110 xxxx
// ITU-T Rec. H.262 | ISO/IEC 13818-2 or ISO/IEC 11172-2 or ISO/IEC
// 14496-2 video stream number xxxx
// ((stream_id >> 4) & 0x0f) == SrsTsPESStreamIdVideo
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if (pprint->can_print()) {
srs_trace("<- " SRS_CONSTS_LOG_STREAM_CASTER " mpegts: got %s age=%d stream=%s, dts=%" PRId64 ", pts=%" PRId64 ", size=%d, us=%d, cc=%d, sid=%#x(%s-%d)",
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(msg->channel->apply == SrsTsPidApplyVideo)? "Video":"Audio", pprint->age(), srs_ts_stream2string(msg->channel->stream).c_str(),
msg->dts, msg->pts, msg->payload->length(), msg->packet->payload_unit_start_indicator, msg->continuity_counter, msg->sid,
msg->is_audio()? "A":msg->is_video()? "V":"N", msg->stream_number());
}
// When the audio SID is private stream 1, we use common audio.
// @see https://github.com/ossrs/srs/issues/740
if (msg->channel->apply == SrsTsPidApplyAudio && msg->sid == SrsTsPESStreamIdPrivateStream1) {
msg->sid = SrsTsPESStreamIdAudioCommon;
}
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// when not audio/video, or not adts/annexb format, donot support.
if (msg->stream_number() != 0) {
return srs_error_new(ERROR_STREAM_CASTER_TS_ES, "ts: unsupported stream format, sid=%#x(%s-%d)",
msg->sid, msg->is_audio()? "A":msg->is_video()? "V":"N", msg->stream_number());
}
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// check supported codec
if (msg->channel->stream != SrsTsStreamVideoH264 && msg->channel->stream != SrsTsStreamAudioAAC) {
return srs_error_new(ERROR_STREAM_CASTER_TS_CODEC, "ts: unsupported stream codec=%d", msg->channel->stream);
}
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// parse the stream.
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SrsBuffer avs(msg->payload->bytes(), msg->payload->length());
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// publish audio or video.
if (msg->channel->stream == SrsTsStreamVideoH264) {
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if ((err = on_ts_video(msg, &avs)) != srs_success) {
return srs_error_wrap(err, "ts: consume video");
}
}
if (msg->channel->stream == SrsTsStreamAudioAAC) {
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if ((err = on_ts_audio(msg, &avs)) != srs_success) {
return srs_error_wrap(err, "ts: consume audio");
}
}
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// TODO: FIXME: implements it.
return err;
}
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srs_error_t SrsMpegtsOverUdp::on_ts_video(SrsTsMessage* msg, SrsBuffer* avs)
{
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srs_error_t err = srs_success;
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// ensure rtmp connected.
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if ((err = connect()) != srs_success) {
return srs_error_wrap(err, "connect");
}
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// ts tbn to flv tbn.
uint32_t dts = (uint32_t)(msg->dts / 90);
uint32_t pts = (uint32_t)(msg->dts / 90);
// send each frame.
while (!avs->empty()) {
char* frame = NULL;
int frame_size = 0;
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if ((err = avc->annexb_demux(avs, &frame, &frame_size)) != srs_success) {
return srs_error_wrap(err, "demux annexb");
}
// 5bits, 7.3.1 NAL unit syntax,
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// ISO_IEC_14496-10-AVC-2003.pdf, page 44.
// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
// ignore the nalu type sps(7), pps(8), aud(9)
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if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
continue;
}
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// for sps
if (avc->is_sps(frame, frame_size)) {
std::string sps;
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if ((err = avc->sps_demux(frame, frame_size, sps)) != srs_success) {
return srs_error_wrap(err, "demux sps");
}
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if (h264_sps == sps) {
continue;
}
h264_sps_changed = true;
h264_sps = sps;
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if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
return srs_error_wrap(err, "write sps/pps");
}
continue;
}
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// for pps
if (avc->is_pps(frame, frame_size)) {
std::string pps;
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if ((err = avc->pps_demux(frame, frame_size, pps)) != srs_success) {
return srs_error_wrap(err, "demux pps");
}
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if (h264_pps == pps) {
continue;
}
h264_pps_changed = true;
h264_pps = pps;
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if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
return srs_error_wrap(err, "write sps/pps");
}
continue;
}
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// ibp frame.
// TODO: FIXME: we should group all frames to a rtmp/flv message from one ts message.
srs_info("mpegts: demux avc ibp frame size=%d, dts=%d", frame_size, dts);
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if ((err = write_h264_ipb_frame(frame, frame_size, dts, pts)) != srs_success) {
return srs_error_wrap(err, "write frame");
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}
}
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return err;
}
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srs_error_t SrsMpegtsOverUdp::write_h264_sps_pps(uint32_t dts, uint32_t pts)
{
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srs_error_t err = srs_success;
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// TODO: FIMXE: there exists bug, see following comments.
// when sps or pps changed, update the sequence header,
// for the pps maybe not changed while sps changed.
// so, we must check when each video ts message frame parsed.
if (!h264_sps_changed || !h264_pps_changed) {
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return err;
}
// h264 raw to h264 packet.
std::string sh;
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if ((err = avc->mux_sequence_header(h264_sps, h264_pps, dts, pts, sh)) != srs_success) {
return srs_error_wrap(err, "mux sequence header");
}
// h264 packet to flv packet.
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int8_t frame_type = SrsVideoAvcFrameTypeKeyFrame;
int8_t avc_packet_type = SrsVideoAvcFrameTraitSequenceHeader;
char* flv = NULL;
int nb_flv = 0;
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if ((err = avc->mux_avc2flv(sh, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
return srs_error_wrap(err, "avc to flv");
}
// the timestamp in rtmp message header is dts.
uint32_t timestamp = dts;
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if ((err = rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv)) != srs_success) {
return srs_error_wrap(err, "write packet");
}
// reset sps and pps.
h264_sps_changed = false;
h264_pps_changed = false;
h264_sps_pps_sent = true;
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return err;
}
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srs_error_t SrsMpegtsOverUdp::write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts)
{
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srs_error_t err = srs_success;
// when sps or pps not sent, ignore the packet.
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// @see https://github.com/ossrs/srs/issues/203
if (!h264_sps_pps_sent) {
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return srs_error_new(ERROR_H264_DROP_BEFORE_SPS_PPS, "drop sps/pps");
}
// 5bits, 7.3.1 NAL unit syntax,
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// ISO_IEC_14496-10-AVC-2003.pdf, page 44.
// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
// for IDR frame, the frame is keyframe.
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SrsVideoAvcFrameType frame_type = SrsVideoAvcFrameTypeInterFrame;
if (nal_unit_type == SrsAvcNaluTypeIDR) {
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frame_type = SrsVideoAvcFrameTypeKeyFrame;
}
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std::string ibp;
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if ((err = avc->mux_ipb_frame(frame, frame_size, ibp)) != srs_success) {
return srs_error_wrap(err, "mux frame");
}
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int8_t avc_packet_type = SrsVideoAvcFrameTraitNALU;
char* flv = NULL;
int nb_flv = 0;
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if ((err = avc->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
return srs_error_wrap(err, "mux avc to flv");
}
// the timestamp in rtmp message header is dts.
uint32_t timestamp = dts;
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return rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv);
}
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srs_error_t SrsMpegtsOverUdp::on_ts_audio(SrsTsMessage* msg, SrsBuffer* avs)
{
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srs_error_t err = srs_success;
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// ensure rtmp connected.
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if ((err = connect()) != srs_success) {
return srs_error_wrap(err, "connect");
}
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// ts tbn to flv tbn.
uint32_t dts = (uint32_t)(msg->dts / 90);
// send each frame.
while (!avs->empty()) {
char* frame = NULL;
int frame_size = 0;
SrsRawAacStreamCodec codec;
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if ((err = aac->adts_demux(avs, &frame, &frame_size, codec)) != srs_success) {
return srs_error_wrap(err, "demux adts");
}
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// ignore invalid frame,
// * atleast 1bytes for aac to decode the data.
if (frame_size <= 0) {
continue;
}
srs_info("mpegts: demux aac frame size=%d, dts=%d", frame_size, dts);
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// generate sh.
if (aac_specific_config.empty()) {
std::string sh;
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if ((err = aac->mux_sequence_header(&codec, sh)) != srs_success) {
return srs_error_wrap(err, "mux sequence header");
}
aac_specific_config = sh;
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codec.aac_packet_type = 0;
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if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != srs_success) {
return srs_error_wrap(err, "write raw audio frame");
}
}
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// audio raw data.
codec.aac_packet_type = 1;
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if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
return srs_error_wrap(err, "write audio raw frame");
}
}
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return err;
}
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srs_error_t SrsMpegtsOverUdp::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts)
{
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srs_error_t err = srs_success;
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char* data = NULL;
int size = 0;
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if ((err = aac->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
return srs_error_wrap(err, "mux aac to flv");
}
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return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
}
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srs_error_t SrsMpegtsOverUdp::rtmp_write_packet(char type, uint32_t timestamp, char* data, int size)
{
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srs_error_t err = srs_success;
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if ((err = connect()) != srs_success) {
return srs_error_wrap(err, "connect");
}
SrsSharedPtrMessage* msg = NULL;
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if ((err = srs_rtmp_create_msg(type, timestamp, data, size, sdk->sid(), &msg)) != srs_success) {
return srs_error_wrap(err, "create message");
}
srs_assert(msg);
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// push msg to queue.
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if ((err = queue->push(msg)) != srs_success) {
return srs_error_wrap(err, "push to queue");
}
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// for all ready msg, dequeue and send out.
for (;;) {
if ((msg = queue->dequeue()) == NULL) {
break;
}
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if (pprint->can_print()) {
srs_trace("mpegts: send msg %s age=%d, dts=%" PRId64 ", size=%d",
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msg->is_audio()? "A":msg->is_video()? "V":"N", pprint->age(), msg->timestamp, msg->size);
}
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// send out encoded msg.
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if ((err = sdk->send_and_free_message(msg)) != srs_success) {
close();
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return srs_error_wrap(err, "send messages");
}
}
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return err;
}
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srs_error_t SrsMpegtsOverUdp::connect()
{
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srs_error_t err = srs_success;
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// Ignore when connected.
if (sdk) {
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return err;
}
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int64_t cto = SRS_CONSTS_RTMP_TMMS;
int64_t sto = SRS_CONSTS_RTMP_PULSE_TMMS;
sdk = new SrsSimpleRtmpClient(output, cto, sto);
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if ((err = sdk->connect()) != srs_success) {
close();
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return srs_error_wrap(err, "connect %s failed, cto=%" PRId64 ", sto=%" PRId64, output.c_str(), cto, sto);
}
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if ((err = sdk->publish()) != srs_success) {
close();
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return srs_error_wrap(err, "publish");
}
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return err;
}
void SrsMpegtsOverUdp::close()
{
srs_freep(sdk);
}
#endif