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srs/trunk/src/app/srs_app_rtc_source.cpp
winlin f33a62e3bd Remove dead code
2020-07-15 13:56:46 +08:00

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/**
* The MIT License (MIT)
*
* Copyright (c) 2013-2020 John
*
* 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_rtc_source.hpp>
#include <srs_app_conn.hpp>
#include <srs_rtmp_stack.hpp>
#include <srs_app_config.hpp>
#include <srs_app_source.hpp>
#include <srs_kernel_flv.hpp>
#include <srs_kernel_codec.hpp>
#include <srs_rtmp_msg_array.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_protocol_format.hpp>
#include <srs_kernel_buffer.hpp>
#include <srs_kernel_rtc_rtp.hpp>
#include <srs_core_autofree.hpp>
#include <srs_app_rtc_queue.hpp>
#include <srs_app_rtc_conn.hpp>
#ifdef SRS_FFMPEG_FIT
#include <srs_app_rtc_codec.hpp>
#endif
const int kChannel = 2;
const int kSamplerate = 48000;
// An AAC packet may be transcoded to many OPUS packets.
const int kMaxOpusPackets = 8;
// The max size for each OPUS packet.
const int kMaxOpusPacketSize = 4096;
// The RTP payload max size, reserved some paddings for SRTP as such:
// kRtpPacketSize = kRtpMaxPayloadSize + paddings
// For example, if kRtpPacketSize is 1500, recommend to set kRtpMaxPayloadSize to 1400,
// which reserves 100 bytes for SRTP or paddings.
const int kRtpMaxPayloadSize = kRtpPacketSize - 200;
using namespace std;
// TODO: Add this function into SrsRtpMux class.
srs_error_t aac_raw_append_adts_header(SrsSharedPtrMessage* shared_audio, SrsFormat* format, char** pbuf, int* pnn_buf)
{
srs_error_t err = srs_success;
if (format->is_aac_sequence_header()) {
return err;
}
if (format->audio->nb_samples != 1) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "adts");
}
int nb_buf = format->audio->samples[0].size + 7;
char* buf = new char[nb_buf];
SrsBuffer stream(buf, nb_buf);
// TODO: Add comment.
stream.write_1bytes(0xFF);
stream.write_1bytes(0xF9);
stream.write_1bytes(((format->acodec->aac_object - 1) << 6) | ((format->acodec->aac_sample_rate & 0x0F) << 2) | ((format->acodec->aac_channels & 0x04) >> 2));
stream.write_1bytes(((format->acodec->aac_channels & 0x03) << 6) | ((nb_buf >> 11) & 0x03));
stream.write_1bytes((nb_buf >> 3) & 0xFF);
stream.write_1bytes(((nb_buf & 0x07) << 5) | 0x1F);
stream.write_1bytes(0xFC);
stream.write_bytes(format->audio->samples[0].bytes, format->audio->samples[0].size);
*pbuf = buf;
*pnn_buf = nb_buf;
return err;
}
uint64_t SrsNtp::kMagicNtpFractionalUnit = 1ULL << 32;
SrsNtp::SrsNtp()
{
system_ms_ = 0;
ntp_ = 0;
ntp_second_ = 0;
ntp_fractions_ = 0;
}
SrsNtp::~SrsNtp()
{
}
SrsNtp SrsNtp::from_time_ms(uint64_t ms)
{
SrsNtp srs_ntp;
srs_ntp.system_ms_ = ms;
srs_ntp.ntp_second_ = ms / 1000;
srs_ntp.ntp_fractions_ = (static_cast<double>(ms % 1000 / 1000.0)) * kMagicNtpFractionalUnit;
srs_ntp.ntp_ = (static_cast<uint64_t>(srs_ntp.ntp_second_) << 32) | srs_ntp.ntp_fractions_;
return srs_ntp;
}
SrsNtp SrsNtp::to_time_ms(uint64_t ntp)
{
SrsNtp srs_ntp;
srs_ntp.ntp_ = ntp;
srs_ntp.ntp_second_ = (ntp & 0xFFFFFFFF00000000ULL) >> 32;
srs_ntp.ntp_fractions_ = (ntp & 0x00000000FFFFFFFFULL);
srs_ntp.system_ms_ = (static_cast<uint64_t>(srs_ntp.ntp_second_) * 1000) +
(static_cast<double>(static_cast<uint64_t>(srs_ntp.ntp_fractions_) * 1000.0) / kMagicNtpFractionalUnit);
return srs_ntp;
}
SrsRtcConsumer::SrsRtcConsumer(SrsRtcStream* s)
{
source = s;
should_update_source_id = false;
mw_wait = srs_cond_new();
mw_min_msgs = 0;
mw_waiting = false;
}
SrsRtcConsumer::~SrsRtcConsumer()
{
source->on_consumer_destroy(this);
vector<SrsRtpPacket2*>::iterator it;
for (it = queue.begin(); it != queue.end(); ++it) {
SrsRtpPacket2* pkt = *it;
srs_freep(pkt);
}
srs_cond_destroy(mw_wait);
}
void SrsRtcConsumer::update_source_id()
{
should_update_source_id = true;
}
srs_error_t SrsRtcConsumer::enqueue(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
queue.push_back(pkt);
if (mw_waiting) {
if ((int)queue.size() > mw_min_msgs) {
srs_cond_signal(mw_wait);
mw_waiting = false;
return err;
}
}
return err;
}
srs_error_t SrsRtcConsumer::dump_packets(std::vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
if (should_update_source_id) {
srs_trace("update source_id=%s[%s]", source->source_id().c_str(), source->source_id().c_str());
should_update_source_id = false;
}
queue.swap(pkts);
return err;
}
void SrsRtcConsumer::wait(int nb_msgs)
{
mw_min_msgs = nb_msgs;
// when duration ok, signal to flush.
if ((int)queue.size() > mw_min_msgs) {
return;
}
// the enqueue will notify this cond.
mw_waiting = true;
// use cond block wait for high performance mode.
srs_cond_wait(mw_wait);
}
SrsRtcStreamManager::SrsRtcStreamManager()
{
lock = NULL;
}
SrsRtcStreamManager::~SrsRtcStreamManager()
{
srs_mutex_destroy(lock);
}
srs_error_t SrsRtcStreamManager::fetch_or_create(SrsRequest* r, SrsRtcStream** pps)
{
srs_error_t err = srs_success;
// Lazy create lock, because ST is not ready in SrsRtcStreamManager constructor.
if (!lock) {
lock = srs_mutex_new();
}
// Use lock to protect coroutine switch.
// @bug https://github.com/ossrs/srs/issues/1230
SrsLocker(lock);
SrsRtcStream* source = NULL;
if ((source = fetch(r)) != NULL) {
*pps = source;
return err;
}
string stream_url = r->get_stream_url();
string vhost = r->vhost;
// should always not exists for create a source.
srs_assert (pool.find(stream_url) == pool.end());
srs_trace("new source, stream_url=%s", stream_url.c_str());
source = new SrsRtcStream();
if ((err = source->initialize(r)) != srs_success) {
return srs_error_wrap(err, "init source %s", r->get_stream_url().c_str());
}
pool[stream_url] = source;
*pps = source;
return err;
}
SrsRtcStream* SrsRtcStreamManager::fetch(SrsRequest* r)
{
SrsRtcStream* source = NULL;
string stream_url = r->get_stream_url();
if (pool.find(stream_url) == pool.end()) {
return NULL;
}
source = pool[stream_url];
// we always update the request of resource,
// for origin auth is on, the token in request maybe invalid,
// and we only need to update the token of request, it's simple.
source->update_auth(r);
return source;
}
SrsRtcStreamManager* _srs_rtc_sources = new SrsRtcStreamManager();
ISrsRtcPublishStream::ISrsRtcPublishStream()
{
}
ISrsRtcPublishStream::~ISrsRtcPublishStream()
{
}
SrsRtcStream::SrsRtcStream()
{
_can_publish = true;
publish_stream_ = NULL;
req = NULL;
#ifdef SRS_FFMPEG_FIT
bridger_ = new SrsRtcFromRtmpBridger(this);
#else
bridger_ = new SrsRtcDummyBridger();
#endif
stream_desc_ = NULL;
}
SrsRtcStream::~SrsRtcStream()
{
// never free the consumers,
// for all consumers are auto free.
consumers.clear();
srs_freep(req);
srs_freep(bridger_);
srs_freep(stream_desc_);
}
srs_error_t SrsRtcStream::initialize(SrsRequest* r)
{
srs_error_t err = srs_success;
req = r->copy();
#ifdef SRS_FFMPEG_FIT
SrsRtcFromRtmpBridger* bridger = dynamic_cast<SrsRtcFromRtmpBridger*>(bridger_);
if ((err = bridger->initialize(req)) != srs_success) {
return srs_error_wrap(err, "bridge initialize");
}
#endif
return err;
}
void SrsRtcStream::update_auth(SrsRequest* r)
{
req->update_auth(r);
}
srs_error_t SrsRtcStream::on_source_id_changed(SrsContextId id)
{
srs_error_t err = srs_success;
if (!_source_id.compare(id)) {
return err;
}
if (_pre_source_id.empty()) {
_pre_source_id = id;
} else if (_pre_source_id.compare(_source_id)) {
_pre_source_id = _source_id;
}
_source_id = id;
// notice all consumer
std::vector<SrsRtcConsumer*>::iterator it;
for (it = consumers.begin(); it != consumers.end(); ++it) {
SrsRtcConsumer* consumer = *it;
consumer->update_source_id();
}
return err;
}
SrsContextId SrsRtcStream::source_id()
{
return _source_id;
}
SrsContextId SrsRtcStream::pre_source_id()
{
return _pre_source_id;
}
ISrsSourceBridger* SrsRtcStream::bridger()
{
return bridger_;
}
srs_error_t SrsRtcStream::create_consumer(SrsRtcConsumer*& consumer)
{
srs_error_t err = srs_success;
consumer = new SrsRtcConsumer(this);
consumers.push_back(consumer);
// TODO: FIXME: Implements edge cluster.
return err;
}
srs_error_t SrsRtcStream::consumer_dumps(SrsRtcConsumer* consumer, bool ds, bool dm, bool dg)
{
srs_error_t err = srs_success;
// print status.
srs_trace("create consumer, no gop cache");
return err;
}
void SrsRtcStream::on_consumer_destroy(SrsRtcConsumer* consumer)
{
std::vector<SrsRtcConsumer*>::iterator it;
it = std::find(consumers.begin(), consumers.end(), consumer);
if (it != consumers.end()) {
consumers.erase(it);
}
}
bool SrsRtcStream::can_publish(bool is_edge)
{
return _can_publish;
}
srs_error_t SrsRtcStream::on_publish()
{
srs_error_t err = srs_success;
// update the request object.
srs_assert(req);
_can_publish = false;
// whatever, the publish thread is the source or edge source,
// save its id to srouce id.
if ((err = on_source_id_changed(_srs_context->get_id())) != srs_success) {
return srs_error_wrap(err, "source id change");
}
// TODO: FIXME: Handle by statistic.
return err;
}
void SrsRtcStream::on_unpublish()
{
// ignore when already unpublished.
if (_can_publish) {
return;
}
srs_trace("cleanup when unpublish");
_can_publish = true;
_source_id = SrsContextId();
// TODO: FIXME: Handle by statistic.
}
ISrsRtcPublishStream* SrsRtcStream::publish_stream()
{
return publish_stream_;
}
void SrsRtcStream::set_publish_stream(ISrsRtcPublishStream* v)
{
publish_stream_ = v;
}
srs_error_t SrsRtcStream::on_rtp(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(pkt->copy())) != srs_success) {
return srs_error_wrap(err, "consume message");
}
}
return err;
}
void SrsRtcStream::set_stream_desc(SrsRtcStreamDescription* stream_desc)
{
srs_freep(stream_desc_);
stream_desc_ = stream_desc->copy();
}
std::vector<SrsRtcTrackDescription*> SrsRtcStream::get_track_desc(std::string type, std::string media_name)
{
std::vector<SrsRtcTrackDescription*> track_descs;
if (!stream_desc_) {
return track_descs;
}
if (type == "audio") {
if (stream_desc_->audio_track_desc_->media_->name_ == media_name) {
track_descs.push_back(stream_desc_->audio_track_desc_);
}
}
if (type == "video") {
std::vector<SrsRtcTrackDescription*>::iterator it = stream_desc_->video_track_descs_.begin();
while (it != stream_desc_->video_track_descs_.end() ){
track_descs.push_back(*it);
++it;
}
}
return track_descs;
}
#ifdef SRS_FFMPEG_FIT
SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcStream* source)
{
req = NULL;
source_ = source;
format = new SrsRtmpFormat();
codec = new SrsAudioRecode(kChannel, kSamplerate);
discard_aac = false;
discard_bframe = false;
merge_nalus = false;
meta = new SrsMetaCache();
audio_timestamp = 0;
audio_sequence = 0;
video_sequence = 0;
}
SrsRtcFromRtmpBridger::~SrsRtcFromRtmpBridger()
{
srs_freep(format);
srs_freep(codec);
srs_freep(meta);
}
srs_error_t SrsRtcFromRtmpBridger::initialize(SrsRequest* r)
{
srs_error_t err = srs_success;
req = r;
if ((err = format->initialize()) != srs_success) {
return srs_error_wrap(err, "format initialize");
}
if ((err = codec->initialize()) != srs_success) {
return srs_error_wrap(err, "init codec");
}
// TODO: FIXME: Support reload.
discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
merge_nalus = _srs_config->get_rtc_server_merge_nalus();
srs_trace("RTC bridge from RTMP, discard_aac=%d, discard_bframe=%d, merge_nalus=%d",
discard_aac, discard_bframe, merge_nalus);
return err;
}
srs_error_t SrsRtcFromRtmpBridger::on_publish()
{
srs_error_t err = srs_success;
// TODO: FIXME: Should sync with bridger?
if ((err = source_->on_publish()) != srs_success) {
return srs_error_wrap(err, "source publish");
}
// Reset the metadata cache, to make VLC happy when disable/enable stream.
// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
meta->clear();
return err;
}
void SrsRtcFromRtmpBridger::on_unpublish()
{
// TODO: FIXME: Should sync with bridger?
source_->on_unpublish();
// Reset the metadata cache, to make VLC happy when disable/enable stream.
// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
meta->update_previous_vsh();
meta->update_previous_ash();
}
srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
// TODO: FIXME: Support parsing OPUS for RTC.
if ((err = format->on_audio(msg)) != srs_success) {
return srs_error_wrap(err, "format consume audio");
}
// Ignore if no format->acodec, it means the codec is not parsed, or unknown codec.
// @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474
if (!format->acodec) {
return err;
}
// ts support audio codec: aac/mp3
SrsAudioCodecId acodec = format->acodec->id;
if (acodec != SrsAudioCodecIdAAC && acodec != SrsAudioCodecIdMP3) {
return err;
}
// When drop aac audio packet, never transcode.
if (discard_aac && acodec == SrsAudioCodecIdAAC) {
return err;
}
// ignore sequence header
srs_assert(format->audio);
char* adts_audio = NULL;
int nn_adts_audio = 0;
// TODO: FIXME: Reserve 7 bytes header when create shared message.
if ((err = aac_raw_append_adts_header(msg, format, &adts_audio, &nn_adts_audio)) != srs_success) {
return srs_error_wrap(err, "aac append header");
}
if (adts_audio) {
err = transcode(adts_audio, nn_adts_audio);
srs_freep(adts_audio);
}
return err;
}
srs_error_t SrsRtcFromRtmpBridger::transcode(char* adts_audio, int nn_adts_audio)
{
srs_error_t err = srs_success;
// Opus packet cache.
static char* opus_payloads[kMaxOpusPackets];
static bool initialized = false;
if (!initialized) {
initialized = true;
static char opus_packets_cache[kMaxOpusPackets][kMaxOpusPacketSize];
opus_payloads[0] = &opus_packets_cache[0][0];
for (int i = 1; i < kMaxOpusPackets; i++) {
opus_payloads[i] = opus_packets_cache[i];
}
}
// Transcode an aac packet to many opus packets.
SrsSample aac;
aac.bytes = adts_audio;
aac.size = nn_adts_audio;
int nn_opus_packets = 0;
int opus_sizes[kMaxOpusPackets];
if ((err = codec->transcode(&aac, opus_payloads, opus_sizes, nn_opus_packets)) != srs_success) {
return srs_error_wrap(err, "recode error");
}
// Save OPUS packets in shared message.
if (nn_opus_packets <= 0) {
return err;
}
int nn_max_extra_payload = 0;
for (int i = 0; i < nn_opus_packets; i++) {
char* data = (char*)opus_payloads[i];
int size = (int)opus_sizes[i];
// TODO: FIXME: Use it to padding audios.
nn_max_extra_payload = srs_max(nn_max_extra_payload, size);
SrsRtpPacket2* pkt = NULL;
SrsAutoFree(SrsRtpPacket2, pkt);
if ((err = package_opus(data, size, &pkt)) != srs_success) {
return srs_error_wrap(err, "package opus");
}
if ((err = source_->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "consume opus");
}
}
return err;
}
srs_error_t SrsRtcFromRtmpBridger::package_opus(char* data, int size, SrsRtpPacket2** ppkt)
{
srs_error_t err = srs_success;
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeAudio;
pkt->header.set_marker(true);
pkt->header.set_sequence(audio_sequence++);
pkt->header.set_timestamp(audio_timestamp);
// TODO: FIXME: Why 960? Need Refactoring?
audio_timestamp += 960;
SrsRtpRawPayload* raw = new SrsRtpRawPayload();
pkt->payload = raw;
raw->payload = new char[size];
raw->nn_payload = size;
memcpy(raw->payload, data, size);
pkt->shared_msg = new SrsSharedPtrMessage();
pkt->shared_msg->wrap(raw->payload, size);
*ppkt = pkt;
return err;
}
srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
// cache the sequence header if h264
bool is_sequence_header = SrsFlvVideo::sh(msg->payload, msg->size);
if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
return srs_error_wrap(err, "meta update video");
}
if ((err = format->on_video(msg)) != srs_success) {
return srs_error_wrap(err, "format consume video");
}
bool has_idr = false;
vector<SrsSample*> samples;
if ((err = filter(msg, format, has_idr, samples)) != srs_success) {
return srs_error_wrap(err, "filter video");
}
int nn_samples = (int)samples.size();
// Well, for each IDR, we append a SPS/PPS before it, which is packaged in STAP-A.
if (has_idr) {
SrsRtpPacket2* pkt = NULL;
SrsAutoFree(SrsRtpPacket2, pkt);
if ((err = package_stap_a(source_, msg, &pkt)) != srs_success) {
return srs_error_wrap(err, "package stap-a");
}
if ((err = source_->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "consume sps/pps");
}
}
// If merge Nalus, we pcakges all NALUs(samples) as one NALU, in a RTP or FUA packet.
vector<SrsRtpPacket2*> pkts;
if (merge_nalus && nn_samples > 1) {
if ((err = package_nalus(msg, samples, pkts)) != srs_success) {
return srs_error_wrap(err, "package nalus as one");
}
} else {
// By default, we package each NALU(sample) to a RTP or FUA packet.
for (int i = 0; i < nn_samples; i++) {
SrsSample* sample = samples[i];
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
if (sample->size <= kRtpMaxPayloadSize) {
if ((err = package_single_nalu(msg, sample, pkts)) != srs_success) {
return srs_error_wrap(err, "package single nalu");
}
} else {
if ((err = package_fu_a(msg, sample, kRtpMaxPayloadSize, pkts)) != srs_success) {
return srs_error_wrap(err, "package fu-a");
}
}
}
}
if (pkts.size() > 0) {
pkts.back()->header.set_marker(true);
}
return consume_packets(pkts);
}
srs_error_t SrsRtcFromRtmpBridger::filter(SrsSharedPtrMessage* msg, SrsFormat* format, bool& has_idr, vector<SrsSample*>& samples)
{
srs_error_t err = srs_success;
// If IDR, we will insert SPS/PPS before IDR frame.
if (format->video && format->video->has_idr) {
has_idr = true;
}
// Update samples to shared frame.
for (int i = 0; i < format->video->nb_samples; ++i) {
SrsSample* sample = &format->video->samples[i];
// Because RTC does not support B-frame, so we will drop them.
// TODO: Drop B-frame in better way, which not cause picture corruption.
if (discard_bframe) {
if ((err = sample->parse_bframe()) != srs_success) {
return srs_error_wrap(err, "parse bframe");
}
if (sample->bframe) {
continue;
}
}
samples.push_back(sample);
}
return err;
}
srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcStream* source, SrsSharedPtrMessage* msg, SrsRtpPacket2** ppkt)
{
srs_error_t err = srs_success;
SrsFormat* format = meta->vsh_format();
if (!format || !format->vcodec) {
return err;
}
// Note that the sps/pps may change, so we should copy it.
const vector<char>& sps = format->vcodec->sequenceParameterSetNALUnit;
const vector<char>& pps = format->vcodec->pictureParameterSetNALUnit;
if (sps.empty() || pps.empty()) {
return srs_error_new(ERROR_RTC_RTP_MUXER, "sps/pps empty");
}
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_marker(false);
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpSTAPPayload* stap = new SrsRtpSTAPPayload();
pkt->payload = stap;
uint8_t header = sps[0];
stap->nri = (SrsAvcNaluType)header;
// Copy the SPS/PPS bytes, because it may change.
int size = (int)(sps.size() + pps.size());
char* payload = new char[size];
pkt->shared_msg = new SrsSharedPtrMessage();
pkt->shared_msg->wrap(payload, size);
if (true) {
SrsSample* sample = new SrsSample();
sample->bytes = payload;
sample->size = (int)sps.size();
stap->nalus.push_back(sample);
memcpy(payload, (char*)&sps[0], sps.size());
payload += (int)sps.size();
}
if (true) {
SrsSample* sample = new SrsSample();
sample->bytes = payload;
sample->size = (int)pps.size();
stap->nalus.push_back(sample);
memcpy(payload, (char*)&pps[0], pps.size());
payload += (int)pps.size();
}
*ppkt = pkt;
srs_info("RTC STAP-A seq=%u, sps %d, pps %d bytes", pkt->header.get_sequence(), sps.size(), pps.size());
return err;
}
srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const vector<SrsSample*>& samples, vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
SrsRtpRawNALUs* raw = new SrsRtpRawNALUs();
for (int i = 0; i < (int)samples.size(); i++) {
SrsSample* sample = samples[i];
// We always ignore bframe here, if config to discard bframe,
// the bframe flag will not be set.
if (sample->bframe) {
continue;
}
raw->push_back(sample->copy());
}
// Ignore empty.
int nn_bytes = raw->nb_bytes();
if (nn_bytes <= 0) {
srs_freep(raw);
return err;
}
if (nn_bytes < kRtpMaxPayloadSize) {
// Package NALUs in a single RTP packet.
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
pkt->payload = raw;
pkt->shared_msg = msg->copy();
pkts.push_back(pkt);
} else {
// We must free it, should never use RTP packets to free it,
// because more than one RTP packet will refer to it.
SrsAutoFree(SrsRtpRawNALUs, raw);
// Package NALUs in FU-A RTP packets.
int fu_payload_size = kRtpMaxPayloadSize;
// The first byte is store in FU-A header.
uint8_t header = raw->skip_first_byte();
uint8_t nal_type = header & kNalTypeMask;
int nb_left = nn_bytes - 1;
int num_of_packet = 1 + (nn_bytes - 1) / fu_payload_size;
for (int i = 0; i < num_of_packet; ++i) {
int packet_size = srs_min(nb_left, fu_payload_size);
SrsRtpFUAPayload* fua = new SrsRtpFUAPayload();
if ((err = raw->read_samples(fua->nalus, packet_size)) != srs_success) {
srs_freep(fua);
return srs_error_wrap(err, "read samples %d bytes, left %d, total %d", packet_size, nb_left, nn_bytes);
}
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
fua->nri = (SrsAvcNaluType)header;
fua->nalu_type = (SrsAvcNaluType)nal_type;
fua->start = bool(i == 0);
fua->end = bool(i == num_of_packet - 1);
pkt->payload = fua;
pkt->shared_msg = msg->copy();
pkts.push_back(pkt);
nb_left -= packet_size;
}
}
return err;
}
// Single NAL Unit Packet @see https://tools.ietf.org/html/rfc6184#section-5.6
srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpRawPayload* raw = new SrsRtpRawPayload();
pkt->payload = raw;
raw->payload = sample->bytes;
raw->nn_payload = sample->size;
pkt->shared_msg = msg->copy();
pkts.push_back(pkt);
return err;
}
srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
char* p = sample->bytes + 1;
int nb_left = sample->size - 1;
uint8_t header = sample->bytes[0];
uint8_t nal_type = header & kNalTypeMask;
int num_of_packet = 1 + (sample->size - 1) / fu_payload_size;
for (int i = 0; i < num_of_packet; ++i) {
int packet_size = srs_min(nb_left, fu_payload_size);
SrsRtpPacket2* pkt = new SrsRtpPacket2();
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
pkt->header.set_timestamp(msg->timestamp * 90);
SrsRtpFUAPayload2* fua = new SrsRtpFUAPayload2();
pkt->payload = fua;
fua->nri = (SrsAvcNaluType)header;
fua->nalu_type = (SrsAvcNaluType)nal_type;
fua->start = bool(i == 0);
fua->end = bool(i == num_of_packet - 1);
fua->payload = p;
fua->size = packet_size;
pkt->shared_msg = msg->copy();
pkts.push_back(pkt);
p += packet_size;
nb_left -= packet_size;
}
return err;
}
srs_error_t SrsRtcFromRtmpBridger::consume_packets(vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
// TODO: FIXME: Consume a range of packets.
for (int i = 0; i < (int)pkts.size(); i++) {
SrsRtpPacket2* pkt = pkts[i];
if ((err = source_->on_rtp(pkt)) != srs_success) {
err = srs_error_wrap(err, "consume sps/pps");
break;
}
}
for (int i = 0; i < (int)pkts.size(); i++) {
SrsRtpPacket2* pkt = pkts[i];
srs_freep(pkt);
}
return err;
}
#endif
SrsRtcDummyBridger::SrsRtcDummyBridger()
{
}
SrsRtcDummyBridger::~SrsRtcDummyBridger()
{
}
srs_error_t SrsRtcDummyBridger::on_publish()
{
return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
}
srs_error_t SrsRtcDummyBridger::on_audio(SrsSharedPtrMessage* /*audio*/)
{
return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
}
srs_error_t SrsRtcDummyBridger::on_video(SrsSharedPtrMessage* /*video*/)
{
return srs_error_new(ERROR_RTC_DUMMY_BRIDGER, "no FFmpeg fit");
}
void SrsRtcDummyBridger::on_unpublish()
{
}
SrsCodecPayload::SrsCodecPayload()
{
}
SrsCodecPayload::SrsCodecPayload(uint8_t pt, std::string encode_name, int sample)
{
pt_ = pt;
name_ = encode_name;
sample_ = sample;
}
SrsCodecPayload::~SrsCodecPayload()
{
}
SrsCodecPayload* SrsCodecPayload::copy()
{
SrsCodecPayload* cp = new SrsCodecPayload();
cp->type_ = type_;
cp->pt_ = pt_;
cp->name_ = name_;
cp->sample_ = sample_;
cp->rtcp_fbs_ = rtcp_fbs_;
return cp;
}
SrsMediaPayloadType SrsCodecPayload::generate_media_payload_type()
{
SrsMediaPayloadType media_payload_type(pt_);
media_payload_type.encoding_name_ = name_;
media_payload_type.clock_rate_ = sample_;
media_payload_type.rtcp_fb_ = rtcp_fbs_;
return media_payload_type;
}
SrsVideoPayload::SrsVideoPayload()
{
}
SrsVideoPayload::SrsVideoPayload(uint8_t pt, std::string encode_name, int sample)
:SrsCodecPayload(pt, encode_name, sample)
{
h264_param_.profile_level_id = "";
h264_param_.packetization_mode = "";
h264_param_.level_asymmerty_allow = "";
}
SrsVideoPayload::~SrsVideoPayload()
{
}
SrsVideoPayload* SrsVideoPayload::copy()
{
SrsVideoPayload* cp = new SrsVideoPayload();
cp->type_ = type_;
cp->pt_ = pt_;
cp->name_ = name_;
cp->sample_ = sample_;
cp->rtcp_fbs_ = rtcp_fbs_;
cp->h264_param_ = h264_param_;
return cp;
}
SrsMediaPayloadType SrsVideoPayload::generate_media_payload_type()
{
SrsMediaPayloadType media_payload_type(pt_);
media_payload_type.encoding_name_ = name_;
media_payload_type.clock_rate_ = sample_;
media_payload_type.rtcp_fb_ = rtcp_fbs_;
std::ostringstream format_specific_param;
if (!h264_param_.level_asymmerty_allow.empty()) {
format_specific_param << "level-asymmetry-allowed=" << h264_param_.level_asymmerty_allow;
}
if (!h264_param_.packetization_mode.empty()) {
format_specific_param << ";packetization-mode=" << h264_param_.packetization_mode;
}
if (!h264_param_.profile_level_id.empty()) {
format_specific_param << ";profile-level-id=" << h264_param_.profile_level_id;
}
media_payload_type.format_specific_param_ = format_specific_param.str();
return media_payload_type;
}
srs_error_t SrsVideoPayload::set_h264_param_desc(std::string fmtp)
{
srs_error_t err = srs_success;
std::vector<std::string> vec = split_str(fmtp, ";");
for (size_t i = 0; i < vec.size(); ++i) {
std::vector<std::string> kv = split_str(vec[i], "=");
if (kv.size() == 2) {
if (kv[0] == "profile-level-id") {
h264_param_.profile_level_id = kv[1];
} else if (kv[0] == "packetization-mode") {
// 6.3. Non-Interleaved Mode
// This mode is in use when the value of the OPTIONAL packetization-mode
// media type parameter is equal to 1. This mode SHOULD be supported.
// It is primarily intended for low-delay applications. Only single NAL
// unit packets, STAP-As, and FU-As MAY be used in this mode. STAP-Bs,
// MTAPs, and FU-Bs MUST NOT be used. The transmission order of NAL
// units MUST comply with the NAL unit decoding order.
// @see https://tools.ietf.org/html/rfc6184#section-6.3
h264_param_.packetization_mode = kv[1];
} else if (kv[0] == "level-asymmetry-allowed") {
h264_param_.level_asymmerty_allow = kv[1];
} else {
return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", kv[0].c_str());
}
} else {
return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", vec[i].c_str());
}
}
return err;
}
SrsAudioPayload::SrsAudioPayload()
{
}
SrsAudioPayload::SrsAudioPayload(uint8_t pt, std::string encode_name, int sample, int channel)
:SrsCodecPayload(pt, encode_name, sample)
{
channel_ = channel;
opus_param_.minptime = 0;
opus_param_.use_inband_fec = false;
opus_param_.usedtx = false;
}
SrsAudioPayload::~SrsAudioPayload()
{
}
SrsAudioPayload* SrsAudioPayload::copy()
{
SrsAudioPayload* cp = new SrsAudioPayload();
cp->type_ = type_;
cp->pt_ = pt_;
cp->name_ = name_;
cp->sample_ = sample_;
cp->rtcp_fbs_ = rtcp_fbs_;
cp->channel_ = channel_;
cp->opus_param_ = opus_param_;
return cp;
}
SrsMediaPayloadType SrsAudioPayload::generate_media_payload_type()
{
SrsMediaPayloadType media_payload_type(pt_);
media_payload_type.encoding_name_ = name_;
media_payload_type.clock_rate_ = sample_;
media_payload_type.encoding_param_ = srs_int2str(channel_);
media_payload_type.rtcp_fb_ = rtcp_fbs_;
std::ostringstream format_specific_param;
if (opus_param_.minptime) {
format_specific_param << "minptime=" << opus_param_.minptime;
}
if (opus_param_.use_inband_fec) {
format_specific_param << ";useinbandfec=1";
}
if (opus_param_.usedtx) {
format_specific_param << ";usedtx=1";
}
media_payload_type.format_specific_param_ = format_specific_param.str();
return media_payload_type;
}
srs_error_t SrsAudioPayload::set_opus_param_desc(std::string fmtp)
{
srs_error_t err = srs_success;
std::vector<std::string> vec = split_str(fmtp, ";");
for (size_t i = 0; i < vec.size(); ++i) {
std::vector<std::string> kv = split_str(vec[i], "=");
if (kv.size() == 2) {
if (kv[0] == "minptime") {
opus_param_.minptime = (int)::atol(kv[1].c_str());
} else if (kv[0] == "useinbandfec") {
opus_param_.use_inband_fec = (kv[1] == "1") ? true : false;
} else if (kv[0] == "usedtx") {
opus_param_.usedtx = (kv[1] == "1") ? true : false;
}
} else {
return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid opus param=%s", vec[i].c_str());
}
}
return err;
}
SrsRtcTrackDescription::SrsRtcTrackDescription()
{
ssrc_ = 0;
rtx_ssrc_ = 0;
fec_ssrc_ = 0;
is_active_ = true;
media_ = NULL;
red_ = NULL;
rtx_ = NULL;
ulpfec_ = NULL;
rsfec_ = NULL;
}
SrsRtcTrackDescription::~SrsRtcTrackDescription()
{
srs_freep(media_);
srs_freep(red_);
srs_freep(rtx_);
srs_freep(ulpfec_);
srs_freep(rsfec_);
}
bool SrsRtcTrackDescription::has_ssrc(uint32_t ssrc)
{
if (ssrc == ssrc_ || ssrc == rtx_ssrc_ || ssrc == fec_ssrc_) {
return true;
}
return false;
}
void SrsRtcTrackDescription::add_rtp_extension_desc(int id, std::string uri)
{
extmaps_[id] = uri;
}
void SrsRtcTrackDescription::set_direction(std::string direction)
{
direction_ = direction;
}
void SrsRtcTrackDescription::set_codec_payload(SrsCodecPayload* payload)
{
media_ = payload;
}
void SrsRtcTrackDescription::create_auxiliary_payload(const std::vector<SrsMediaPayloadType> payloads)
{
if (!payloads.size()) {
return;
}
SrsMediaPayloadType payload = payloads.at(0);
if (payload.encoding_name_ == "red"){
srs_freep(red_);
red_ = new SrsCodecPayload(payload.payload_type_, "red", payload.clock_rate_);
} else if (payload.encoding_name_ == "rtx") {
srs_freep(rtx_);
rtx_ = new SrsCodecPayload(payload.payload_type_, "rtx", payload.clock_rate_);
} else if (payload.encoding_name_ == "ulpfec") {
srs_freep(ulpfec_);
ulpfec_ = new SrsCodecPayload(payload.payload_type_, "ulpfec", payload.clock_rate_);
} else if (payload.encoding_name_ == "rsfec") {
srs_freep(rsfec_);
rsfec_ = new SrsCodecPayload(payload.payload_type_, "rsfec", payload.clock_rate_);
}
}
void SrsRtcTrackDescription::set_rtx_ssrc(uint32_t ssrc)
{
rtx_ssrc_ = ssrc;
}
void SrsRtcTrackDescription::set_fec_ssrc(uint32_t ssrc)
{
fec_ssrc_ = ssrc;
}
void SrsRtcTrackDescription::set_mid(std::string mid)
{
mid_ = mid;
}
int SrsRtcTrackDescription::get_rtp_extension_id(std::string uri)
{
for(std::map<int, std::string>::iterator it = extmaps_.begin(); it != extmaps_.end(); ++it) {
if(uri == it->second) {
return it->first;
}
}
return -1;
}
SrsRtcTrackDescription* SrsRtcTrackDescription::copy()
{
SrsRtcTrackDescription* cp = new SrsRtcTrackDescription();
cp->type_ = type_;
cp->id_ = id_;
cp->ssrc_ = ssrc_;
cp->fec_ssrc_ = fec_ssrc_;
cp->rtx_ssrc_ = rtx_ssrc_;
cp->extmaps_ = extmaps_;
cp->direction_ = direction_;
cp->mid_ = mid_;
cp->is_active_ = is_active_;
cp->media_ = media_ ? media_->copy():NULL;
cp->red_ = red_ ? red_->copy():NULL;
cp->rtx_ = rtx_ ? rtx_->copy():NULL;
cp->ulpfec_ = ulpfec_ ? ulpfec_->copy():NULL;
cp->rsfec_ = rsfec_ ? rsfec_->copy():NULL;
return cp;
}
SrsRtcStreamDescription::SrsRtcStreamDescription()
{
audio_track_desc_ = NULL;
}
SrsRtcStreamDescription::~SrsRtcStreamDescription()
{
srs_freep(audio_track_desc_);
for (int i = 0; i < video_track_descs_.size(); ++i) {
srs_freep(video_track_descs_.at(i));
}
video_track_descs_.clear();
}
SrsRtcStreamDescription* SrsRtcStreamDescription::copy()
{
SrsRtcStreamDescription* stream_desc = new SrsRtcStreamDescription();
if (audio_track_desc_) {
stream_desc->audio_track_desc_ = audio_track_desc_->copy();
}
for (int i = 0; i < video_track_descs_.size(); ++i) {
stream_desc->video_track_descs_.push_back(video_track_descs_.at(i)->copy());
}
return stream_desc;
}
SrsRtcTrackDescription* SrsRtcStreamDescription::find_track_description_by_ssrc(uint32_t ssrc)
{
if (audio_track_desc_->has_ssrc(ssrc)) {
return audio_track_desc_;
}
for (int i = 0; i < video_track_descs_.size(); ++i) {
if (video_track_descs_.at(i)->has_ssrc(ssrc)) {
return video_track_descs_.at(i);
}
}
return NULL;
}
SrsRtcRecvTrack::SrsRtcRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc, bool is_audio)
{
session_ = session;
track_desc_ = track_desc->copy();
if (is_audio) {
rtp_queue_ = new SrsRtpRingBuffer(100);
nack_receiver_ = new SrsRtpNackForReceiver(rtp_queue_, 100 * 2 / 3);
} else {
rtp_queue_ = new SrsRtpRingBuffer(1000);
nack_receiver_ = new SrsRtpNackForReceiver(rtp_queue_, 1000 * 2 / 3);
}
}
SrsRtcRecvTrack::~SrsRtcRecvTrack()
{
srs_freep(rtp_queue_);
srs_freep(nack_receiver_);
srs_freep(track_desc_);
}
bool SrsRtcRecvTrack::has_ssrc(uint32_t ssrc)
{
if (track_desc_) {
return track_desc_->has_ssrc(ssrc);
}
return false;
}
void SrsRtcRecvTrack::update_rtt(int rtt)
{
if (nack_receiver_) {
nack_receiver_->update_rtt(rtt);
}
}
void SrsRtcRecvTrack::update_send_report_time(const SrsNtp& ntp)
{
last_sender_report_ntp = ntp;
last_sender_report_sys_time = srs_update_system_time();;
}
srs_error_t SrsRtcRecvTrack::send_rtcp_rr()
{
srs_error_t err = srs_success;
if (session_) {
return session_->send_rtcp_rr(track_desc_->ssrc_, rtp_queue_, last_sender_report_sys_time, last_sender_report_ntp);
}
return err;
}
srs_error_t SrsRtcRecvTrack::send_rtcp_xr_rrtr()
{
srs_error_t err = srs_success;
if (track_desc_) {
return session_->send_rtcp_xr_rrtr(track_desc_->ssrc_);
}
return err;
}
srs_error_t SrsRtcRecvTrack::on_nack(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
uint32_t ssrc = pkt->header.get_ssrc();
uint16_t seq = pkt->header.get_sequence();
// TODO: check whether is necessary?
nack_receiver_->remove_timeout_packets();
SrsRtpNackInfo* nack_info = nack_receiver_->find(seq);
if (nack_info) {
// seq had been received.
nack_receiver_->remove(seq);
return err;
}
// insert check nack list
uint16_t nack_first = 0, nack_last = 0;
if (!rtp_queue_->update(seq, nack_first, nack_last)) {
srs_warn("too old seq %u, range [%u, %u]", seq, rtp_queue_->begin, rtp_queue_->end);
}
if (srs_rtp_seq_distance(nack_first, nack_last) > 0) {
srs_trace("update seq=%u, nack range [%u, %u]", seq, nack_first, nack_last);
nack_receiver_->insert(nack_first, nack_last);
nack_receiver_->check_queue_size();
}
// insert into video_queue and audio_queue
rtp_queue_->set(seq, pkt->copy());
// send_nack
session_->check_send_nacks(nack_receiver_, ssrc);
return err;
}
srs_error_t SrsRtcRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt)
{
return srs_success;
}
SrsRtcAudioRecvTrack::SrsRtcAudioRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc)
: SrsRtcRecvTrack(session, track_desc, true)
{
}
SrsRtcAudioRecvTrack::~SrsRtcAudioRecvTrack()
{
}
srs_error_t SrsRtcAudioRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
// uint8_t pt = pkt->header.get_payload_type();
// SrsRtcTrackDescription track = rtc_stream_desc_->get_audio_tracks();
// // process red packet.
// if (pt == red_pt) {
// } else if (pt == rtx_pt) { // process rtx_pt.
// // restore retranmission packet.
// } else if (pt == fec_pt) {
// }
if (source) {
if ((err = source->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "source on rtp");
}
}
// For NACK to handle packet.
if ((err = on_nack(pkt)) != srs_success) {
return srs_error_wrap(err, "on nack");
}
return err;
}
SrsRtcVideoRecvTrack::SrsRtcVideoRecvTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc)
: SrsRtcRecvTrack(session, track_desc, false)
{
request_key_frame_ = false;
}
SrsRtcVideoRecvTrack::~SrsRtcVideoRecvTrack()
{
}
srs_error_t SrsRtcVideoRecvTrack::on_rtp(SrsRtcStream* source, SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
pkt->frame_type = SrsFrameTypeVideo;
// TODO: FIXME: add rtp process
if (request_key_frame_) {
// TODO: FIXME: add coroutine to request key frame.
request_key_frame_ = false;
// TODO: FIXME: Check error.
session_->send_rtcp_fb_pli(track_desc_->ssrc_);
}
if (source) {
if ((err = source->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "source on rtp");
}
}
// For NACK to handle packet.
if ((err = on_nack(pkt)) != srs_success) {
return srs_error_wrap(err, "on nack");
}
return err;
}
void SrsRtcVideoRecvTrack::request_keyframe()
{
request_key_frame_ = true;
}
SrsRtcSendTrack::SrsRtcSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc, bool is_audio)
{
session_ = session;
track_desc_ = track_desc->copy();
if (is_audio) {
rtp_queue_ = new SrsRtpRingBuffer(100);
} else {
rtp_queue_ = new SrsRtpRingBuffer(1000);
}
}
SrsRtcSendTrack::~SrsRtcSendTrack()
{
srs_freep(rtp_queue_);
srs_freep(track_desc_);
}
bool SrsRtcSendTrack::has_ssrc(uint32_t ssrc)
{
if (track_desc_) {
return track_desc_->has_ssrc(ssrc);
}
return false;
}
SrsRtpPacket2* SrsRtcSendTrack::fetch_rtp_packet(uint16_t seq)
{
if (rtp_queue_) {
return rtp_queue_->at(seq);
}
return NULL;
}
srs_error_t SrsRtcSendTrack::on_rtp(std::vector<SrsRtpPacket2*>& send_packets, SrsRtpPacket2* pkt)
{
return srs_success;
}
srs_error_t SrsRtcSendTrack::on_rtcp(SrsRtpPacket2* pkt)
{
return srs_success;
}
SrsRtcAudioSendTrack::SrsRtcAudioSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc)
: SrsRtcSendTrack(session, track_desc, true)
{
}
SrsRtcAudioSendTrack::~SrsRtcAudioSendTrack()
{
}
srs_error_t SrsRtcAudioSendTrack::on_rtp(std::vector<SrsRtpPacket2*>& send_packets, SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
pkt->header.set_ssrc(track_desc_->ssrc_);
pkt->header.set_payload_type(track_desc_->media_->pt_);
// Put rtp packet to NACK/ARQ queue
if (true) {
SrsRtpPacket2* nack = pkt->copy();
rtp_queue_->set(nack->header.get_sequence(), nack);
}
send_packets.push_back(pkt);
return err;
}
srs_error_t SrsRtcAudioSendTrack::on_rtcp(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
// process rtcp
return err;
}
SrsRtcVideoSendTrack::SrsRtcVideoSendTrack(SrsRtcConnection* session, SrsRtcTrackDescription* track_desc)
: SrsRtcSendTrack(session, track_desc, false)
{
}
SrsRtcVideoSendTrack::~SrsRtcVideoSendTrack()
{
}
srs_error_t SrsRtcVideoSendTrack::on_rtp(std::vector<SrsRtpPacket2*>& send_packets, SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
pkt->header.set_ssrc(track_desc_->ssrc_);
pkt->header.set_payload_type(track_desc_->media_->pt_);
// Put rtp packet to NACK/ARQ queue
if (true) {
SrsRtpPacket2* nack = pkt->copy();
rtp_queue_->set(nack->header.get_sequence(), nack);
}
send_packets.push_back(pkt);
return err;
}
srs_error_t SrsRtcVideoSendTrack::on_rtcp(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
// process rtcp
return err;
}
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