mirror of https://github.com/ossrs/srs.git
RTC: Extract RTC Source and Consumer
parent
d0d8f676c3
commit
25496b734b
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/**
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* The MIT License (MIT)
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*
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* Copyright (c) 2013-2020 John
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
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* the Software, and to permit persons to whom the Software is furnished to do so,
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* subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
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* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
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* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
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* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
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* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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#include <srs_app_rtc_source.hpp>
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#include <srs_app_conn.hpp>
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#include <srs_rtmp_stack.hpp>
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#include <srs_app_config.hpp>
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#include <srs_app_source.hpp>
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#include <srs_kernel_flv.hpp>
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#include <srs_kernel_codec.hpp>
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#include <srs_rtmp_msg_array.hpp>
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#include <srs_kernel_utility.hpp>
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#include <srs_protocol_format.hpp>
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#include <srs_app_rtc.hpp>
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using namespace std;
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SrsRtcConsumer::SrsRtcConsumer(SrsRtcSource* s, SrsConnection* c)
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{
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source = s;
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conn = c;
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should_update_source_id = false;
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queue = new SrsMessageQueue();
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#ifdef SRS_PERF_QUEUE_COND_WAIT
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mw_wait = srs_cond_new();
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mw_min_msgs = 0;
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mw_duration = 0;
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mw_waiting = false;
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#endif
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}
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SrsRtcConsumer::~SrsRtcConsumer()
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{
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source->on_consumer_destroy(this);
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srs_freep(queue);
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#ifdef SRS_PERF_QUEUE_COND_WAIT
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srs_cond_destroy(mw_wait);
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#endif
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}
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void SrsRtcConsumer::enable_pass_timestamp()
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{
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}
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void SrsRtcConsumer::set_queue_size(srs_utime_t queue_size)
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{
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}
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void SrsRtcConsumer::update_source_id()
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{
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should_update_source_id = true;
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}
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srs_error_t SrsRtcConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag)
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{
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srs_error_t err = srs_success;
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SrsSharedPtrMessage* msg = shared_msg->copy();
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if ((err = queue->enqueue(msg, NULL, true)) != srs_success) {
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return srs_error_wrap(err, "enqueue message");
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}
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#ifdef SRS_PERF_QUEUE_COND_WAIT
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// fire the mw when msgs is enough.
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if (mw_waiting) {
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if (queue->size() > mw_min_msgs) {
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srs_cond_signal(mw_wait);
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mw_waiting = false;
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return err;
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}
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return err;
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}
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#endif
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return err;
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}
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srs_error_t SrsRtcConsumer::dump_packets(SrsMessageArray* msgs, int& count)
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{
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srs_error_t err = srs_success;
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srs_assert(count >= 0);
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srs_assert(msgs->max > 0);
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// the count used as input to reset the max if positive.
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int max = count? srs_min(count, msgs->max) : msgs->max;
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// the count specifies the max acceptable count,
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// here maybe 1+, and we must set to 0 when got nothing.
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count = 0;
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if (should_update_source_id) {
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srs_trace("update source_id=%d[%d]", source->source_id(), source->source_id());
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should_update_source_id = false;
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}
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// pump msgs from queue.
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if ((err = queue->dump_packets(max, msgs->msgs, count, true)) != srs_success) {
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return srs_error_wrap(err, "dump packets");
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}
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return err;
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}
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#ifdef SRS_PERF_QUEUE_COND_WAIT
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void SrsRtcConsumer::wait(int nb_msgs, srs_utime_t msgs_duration)
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{
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mw_min_msgs = nb_msgs;
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mw_duration = msgs_duration;
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srs_utime_t duration = queue->duration();
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bool match_min_msgs = queue->size() > mw_min_msgs;
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// when duration ok, signal to flush.
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if (match_min_msgs && duration > mw_duration) {
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return;
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}
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// the enqueue will notify this cond.
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mw_waiting = true;
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// use cond block wait for high performance mode.
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srs_cond_wait(mw_wait);
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}
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#endif
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SrsRtcSource::SrsRtcSource()
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{
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req = NULL;
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_source_id = _pre_source_id = -1;
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meta = new SrsMetaCache();
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format = new SrsRtmpFormat();
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rtc = new SrsRtc();
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_can_publish = true;
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}
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SrsRtcSource::~SrsRtcSource()
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{
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// never free the consumers,
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// for all consumers are auto free.
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consumers.clear();
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srs_freep(meta);
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srs_freep(format);
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srs_freep(rtc);
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srs_freep(req);
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}
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srs_error_t SrsRtcSource::initialize(SrsRequest* r)
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{
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srs_error_t err = srs_success;
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req = r->copy();
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if ((err = format->initialize()) != srs_success) {
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return srs_error_wrap(err, "format initialize");
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}
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if ((err = rtc->initialize(req)) != srs_success) {
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return srs_error_wrap(err, "rtc initialize");
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}
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return err;
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}
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void SrsRtcSource::update_auth(SrsRequest* r)
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{
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req->update_auth(r);
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}
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srs_error_t SrsRtcSource::on_source_id_changed(int id)
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{
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srs_error_t err = srs_success;
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if (_source_id == id) {
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return err;
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}
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if (_pre_source_id == -1) {
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_pre_source_id = id;
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} else if (_pre_source_id != _source_id) {
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_pre_source_id = _source_id;
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}
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_source_id = id;
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// notice all consumer
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std::vector<SrsRtcConsumer*>::iterator it;
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for (it = consumers.begin(); it != consumers.end(); ++it) {
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SrsRtcConsumer* consumer = *it;
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consumer->update_source_id();
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}
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return err;
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}
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int SrsRtcSource::source_id()
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{
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return _source_id;
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}
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int SrsRtcSource::pre_source_id()
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{
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return _pre_source_id;
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}
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srs_error_t SrsRtcSource::create_consumer(SrsConnection* conn, SrsRtcConsumer*& consumer)
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{
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srs_error_t err = srs_success;
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consumer = new SrsRtcConsumer(this, conn);
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consumers.push_back(consumer);
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// TODO: FIXME: Implements edge cluster.
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return err;
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}
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srs_error_t SrsRtcSource::consumer_dumps(SrsRtcConsumer* consumer, bool ds, bool dm, bool dg)
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{
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srs_error_t err = srs_success;
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srs_utime_t queue_size = _srs_config->get_queue_length(req->vhost);
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consumer->set_queue_size(queue_size);
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// Copy metadata and sequence header to consumer.
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// TODO: FIXME: Maybe should not do this for RTC?
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if ((err = meta->dumps(consumer, true, SrsRtmpJitterAlgorithmOFF, dm, dg)) != srs_success) {
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return srs_error_wrap(err, "meta dumps");
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}
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// print status.
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if (dg) {
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srs_trace("create consumer, queue_size=%.2f", queue_size);
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} else {
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srs_trace("create consumer, ignore gop cache");
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}
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return err;
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}
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void SrsRtcSource::on_consumer_destroy(SrsRtcConsumer* consumer)
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{
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std::vector<SrsRtcConsumer*>::iterator it;
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it = std::find(consumers.begin(), consumers.end(), consumer);
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if (it != consumers.end()) {
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consumers.erase(it);
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}
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}
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bool SrsRtcSource::can_publish(bool is_edge)
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{
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return _can_publish;
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}
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srs_error_t SrsRtcSource::on_publish()
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{
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srs_error_t err = srs_success;
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// update the request object.
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srs_assert(req);
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_can_publish = false;
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if ((err = rtc->on_publish()) != srs_success) {
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return srs_error_wrap(err, "rtc publish");
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}
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// whatever, the publish thread is the source or edge source,
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// save its id to srouce id.
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if ((err = on_source_id_changed(_srs_context->get_id())) != srs_success) {
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return srs_error_wrap(err, "source id change");
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}
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// Reset the metadata cache, to make VLC happy when disable/enable stream.
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// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
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meta->clear();
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// TODO: FIXME: Handle by statistic.
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return err;
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}
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void SrsRtcSource::on_unpublish()
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{
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// ignore when already unpublished.
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if (_can_publish) {
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return;
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}
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rtc->on_unpublish();
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// Reset the metadata cache, to make VLC happy when disable/enable stream.
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// @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
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meta->update_previous_vsh();
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meta->update_previous_ash();
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srs_trace("cleanup when unpublish");
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_can_publish = true;
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_source_id = -1;
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// TODO: FIXME: Handle by statistic.
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}
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SrsMetaCache* SrsRtcSource::cached_meta()
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{
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return meta;
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}
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SrsRtcPublisher* SrsRtcSource::rtc_publisher()
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{
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return rtc_publisher_;
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}
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void SrsRtcSource::set_rtc_publisher(SrsRtcPublisher* v)
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{
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rtc_publisher_ = v;
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}
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srs_error_t SrsRtcSource::on_rtc_audio(SrsSharedPtrMessage* audio)
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{
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// TODO: FIXME: Merge with on_audio.
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// TODO: FIXME: Print key information.
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return on_audio_imp(audio);
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}
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srs_error_t SrsRtcSource::on_video(SrsCommonMessage* shared_video)
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{
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srs_error_t err = srs_success;
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// drop any unknown header video.
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// @see https://github.com/ossrs/srs/issues/421
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if (!SrsFlvVideo::acceptable(shared_video->payload, shared_video->size)) {
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char b0 = 0x00;
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if (shared_video->size > 0) {
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b0 = shared_video->payload[0];
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}
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srs_warn("drop unknown header video, size=%d, bytes[0]=%#x", shared_video->size, b0);
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return err;
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}
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// convert shared_video to msg, user should not use shared_video again.
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// the payload is transfer to msg, and set to NULL in shared_video.
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SrsSharedPtrMessage msg;
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if ((err = msg.create(shared_video)) != srs_success) {
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return srs_error_wrap(err, "create message");
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}
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// directly process the video message.
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return on_video_imp(&msg);
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}
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srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
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{
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srs_error_t err = srs_success;
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bool is_aac_sequence_header = SrsFlvAudio::sh(msg->payload, msg->size);
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bool is_sequence_header = is_aac_sequence_header;
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// whether consumer should drop for the duplicated sequence header.
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bool drop_for_reduce = false;
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if (is_sequence_header && meta->previous_ash() && _srs_config->get_reduce_sequence_header(req->vhost)) {
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if (meta->previous_ash()->size == msg->size) {
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drop_for_reduce = srs_bytes_equals(meta->previous_ash()->payload, msg->payload, msg->size);
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srs_warn("drop for reduce sh audio, size=%d", msg->size);
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}
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}
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// copy to all consumer
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if (!drop_for_reduce) {
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for (int i = 0; i < (int)consumers.size(); i++) {
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SrsRtcConsumer* consumer = consumers.at(i);
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if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
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return srs_error_wrap(err, "consume message");
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}
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}
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}
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// cache the sequence header of aac, or first packet of mp3.
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// for example, the mp3 is used for hls to write the "right" audio codec.
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// TODO: FIXME: to refine the stream info system.
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if (is_aac_sequence_header || !meta->ash()) {
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if ((err = meta->update_ash(msg)) != srs_success) {
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return srs_error_wrap(err, "meta consume audio");
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}
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}
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// when sequence header, donot push to gop cache and adjust the timestamp.
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if (is_sequence_header) {
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return err;
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}
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// if atc, update the sequence header to abs time.
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if (meta->ash()) {
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meta->ash()->timestamp = msg->timestamp;
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}
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if (meta->data()) {
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meta->data()->timestamp = msg->timestamp;
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}
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return err;
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}
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srs_error_t SrsRtcSource::on_video_imp(SrsSharedPtrMessage* msg)
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{
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srs_error_t err = srs_success;
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bool is_sequence_header = SrsFlvVideo::sh(msg->payload, msg->size);
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// user can disable the sps parse to workaround when parse sps failed.
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// @see https://github.com/ossrs/srs/issues/474
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if (is_sequence_header) {
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format->avc_parse_sps = _srs_config->get_parse_sps(req->vhost);
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}
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if ((err = format->on_video(msg)) != srs_success) {
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return srs_error_wrap(err, "format consume video");
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}
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// Parse RTMP message to RTP packets, in FU-A if too large.
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if ((err = rtc->on_video(msg, format)) != srs_success) {
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// TODO: We should support more strategies.
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srs_warn("rtc: ignore video error %s", srs_error_desc(err).c_str());
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srs_error_reset(err);
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rtc->on_unpublish();
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}
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// whether consumer should drop for the duplicated sequence header.
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bool drop_for_reduce = false;
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if (is_sequence_header && meta->previous_vsh() && _srs_config->get_reduce_sequence_header(req->vhost)) {
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if (meta->previous_vsh()->size == msg->size) {
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drop_for_reduce = srs_bytes_equals(meta->previous_vsh()->payload, msg->payload, msg->size);
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srs_warn("drop for reduce sh video, size=%d", msg->size);
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}
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}
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// cache the sequence header if h264
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if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
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return srs_error_wrap(err, "meta update video");
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}
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// copy to all consumer
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if (!drop_for_reduce) {
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for (int i = 0; i < (int)consumers.size(); i++) {
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SrsRtcConsumer* consumer = consumers.at(i);
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if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
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return srs_error_wrap(err, "consume video");
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}
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}
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}
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// when sequence header, donot push to gop cache and adjust the timestamp.
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if (is_sequence_header) {
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return err;
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}
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// if atc, update the sequence header to abs time.
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if (meta->vsh()) {
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meta->vsh()->timestamp = msg->timestamp;
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}
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if (meta->data()) {
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meta->data()->timestamp = msg->timestamp;
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||||
}
|
||||
|
||||
return err;
|
||||
}
|
||||
|
||||
SrsRtcSourceManager::SrsRtcSourceManager()
|
||||
{
|
||||
lock = NULL;
|
||||
}
|
||||
|
||||
SrsRtcSourceManager::~SrsRtcSourceManager()
|
||||
{
|
||||
srs_mutex_destroy(lock);
|
||||
}
|
||||
|
||||
srs_error_t SrsRtcSourceManager::fetch_or_create(SrsRequest* r, SrsRtcSource** pps)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
// Lazy create lock, because ST is not ready in SrsRtcSourceManager constructor.
|
||||
if (!lock) {
|
||||
lock = srs_mutex_new();
|
||||
}
|
||||
|
||||
// Use lock to protect coroutine switch.
|
||||
// @bug https://github.com/ossrs/srs/issues/1230
|
||||
SrsLocker(lock);
|
||||
|
||||
SrsRtcSource* 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 SrsRtcSource();
|
||||
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;
|
||||
}
|
||||
|
||||
SrsRtcSource* SrsRtcSourceManager::fetch(SrsRequest* r)
|
||||
{
|
||||
SrsRtcSource* 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;
|
||||
}
|
||||
|
||||
SrsRtcSourceManager* _srs_rtc_sources = new SrsRtcSourceManager();
|
||||
|
@ -0,0 +1,181 @@
|
||||
/**
|
||||
* 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.
|
||||
*/
|
||||
|
||||
#ifndef SRS_APP_RTC_SOURCE_HPP
|
||||
#define SRS_APP_RTC_SOURCE_HPP
|
||||
|
||||
#include <srs_core.hpp>
|
||||
|
||||
#include <vector>
|
||||
#include <map>
|
||||
|
||||
#include <srs_service_st.hpp>
|
||||
#include <srs_app_source.hpp>
|
||||
|
||||
class SrsRequest;
|
||||
class SrsConnection;
|
||||
class SrsMetaCache;
|
||||
class SrsRtcPublisher;
|
||||
class SrsSharedPtrMessage;
|
||||
class SrsCommonMessage;
|
||||
class SrsMessageArray;
|
||||
class SrsRtcSource;
|
||||
|
||||
class SrsRtcConsumer : public ISrsConsumerQueue
|
||||
{
|
||||
private:
|
||||
SrsRtcSource* source;
|
||||
// The owner connection for debug, maybe NULL.
|
||||
SrsConnection* conn;
|
||||
SrsMessageQueue* queue;
|
||||
// when source id changed, notice all consumers
|
||||
bool should_update_source_id;
|
||||
#ifdef SRS_PERF_QUEUE_COND_WAIT
|
||||
// The cond wait for mw.
|
||||
// @see https://github.com/ossrs/srs/issues/251
|
||||
srs_cond_t mw_wait;
|
||||
bool mw_waiting;
|
||||
int mw_min_msgs;
|
||||
srs_utime_t mw_duration;
|
||||
#endif
|
||||
public:
|
||||
SrsRtcConsumer(SrsRtcSource* s, SrsConnection* c);
|
||||
virtual ~SrsRtcConsumer();
|
||||
public:
|
||||
// Use pass timestamp mode.
|
||||
// TODO: FIXME: Remove it.
|
||||
void enable_pass_timestamp();
|
||||
// Set the size of queue.
|
||||
// TODO: FIXME: Remove it.
|
||||
virtual void set_queue_size(srs_utime_t queue_size);
|
||||
// when source id changed, notice client to print.
|
||||
virtual void update_source_id();
|
||||
// Enqueue an shared ptr message.
|
||||
// @param shared_msg, directly ptr, copy it if need to save it.
|
||||
// @param whether atc, donot use jitter correct if true.
|
||||
// @param ag the algorithm of time jitter.
|
||||
virtual srs_error_t enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag);
|
||||
// Get packets in consumer queue.
|
||||
// @param msgs the msgs array to dump packets to send.
|
||||
// @param count the count in array, intput and output param.
|
||||
// @remark user can specifies the count to get specified msgs; 0 to get all if possible.
|
||||
virtual srs_error_t dump_packets(SrsMessageArray* msgs, int& count);
|
||||
#ifdef SRS_PERF_QUEUE_COND_WAIT
|
||||
// wait for messages incomming, atleast nb_msgs and in duration.
|
||||
// @param nb_msgs the messages count to wait.
|
||||
// @param msgs_duration the messages duration to wait.
|
||||
virtual void wait(int nb_msgs, srs_utime_t msgs_duration);
|
||||
#endif
|
||||
};
|
||||
|
||||
class SrsRtcSource
|
||||
{
|
||||
private:
|
||||
// For publish, it's the publish client id.
|
||||
// For edge, it's the edge ingest id.
|
||||
// when source id changed, for example, the edge reconnect,
|
||||
// invoke the on_source_id_changed() to let all clients know.
|
||||
int _source_id;
|
||||
// previous source id.
|
||||
int _pre_source_id;
|
||||
SrsRequest* req;
|
||||
SrsRtcPublisher* rtc_publisher_;
|
||||
private:
|
||||
// To delivery stream to clients.
|
||||
std::vector<SrsRtcConsumer*> consumers;
|
||||
// The metadata cache.
|
||||
SrsMetaCache* meta;
|
||||
// Whether source is avaiable for publishing.
|
||||
bool _can_publish;
|
||||
private:
|
||||
// The format, codec information.
|
||||
SrsRtmpFormat* format;
|
||||
// rtc handler
|
||||
SrsRtc* rtc;
|
||||
public:
|
||||
SrsRtcSource();
|
||||
virtual ~SrsRtcSource();
|
||||
public:
|
||||
virtual srs_error_t initialize(SrsRequest* r);
|
||||
// Update the authentication information in request.
|
||||
virtual void update_auth(SrsRequest* r);
|
||||
// The source id changed.
|
||||
virtual srs_error_t on_source_id_changed(int id);
|
||||
// Get current source id.
|
||||
virtual int source_id();
|
||||
virtual int pre_source_id();
|
||||
public:
|
||||
// Create consumer
|
||||
// @param consumer, output the create consumer.
|
||||
virtual srs_error_t create_consumer(SrsConnection* conn, SrsRtcConsumer*& consumer);
|
||||
// Dumps packets in cache to consumer.
|
||||
// @param ds, whether dumps the sequence header.
|
||||
// @param dm, whether dumps the metadata.
|
||||
// @param dg, whether dumps the gop cache.
|
||||
virtual srs_error_t consumer_dumps(SrsRtcConsumer* consumer, bool ds = true, bool dm = true, bool dg = true);
|
||||
virtual void on_consumer_destroy(SrsRtcConsumer* consumer);
|
||||
// TODO: FIXME: Remove the param is_edge.
|
||||
virtual bool can_publish(bool is_edge);
|
||||
// When start publish stream.
|
||||
virtual srs_error_t on_publish();
|
||||
// When stop publish stream.
|
||||
virtual void on_unpublish();
|
||||
public:
|
||||
// For RTC, we need to package SPS/PPS(in cached meta) before each IDR.
|
||||
SrsMetaCache* cached_meta();
|
||||
// Get and set the publisher, passed to consumer to process requests such as PLI.
|
||||
SrsRtcPublisher* rtc_publisher();
|
||||
void set_rtc_publisher(SrsRtcPublisher* v);
|
||||
// When got RTC audio message, which is encoded in opus.
|
||||
// TODO: FIXME: Merge with on_audio.
|
||||
srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
|
||||
virtual srs_error_t on_video(SrsCommonMessage* video);
|
||||
private:
|
||||
virtual srs_error_t on_audio_imp(SrsSharedPtrMessage* audio);
|
||||
virtual srs_error_t on_video_imp(SrsSharedPtrMessage* video);
|
||||
};
|
||||
|
||||
class SrsRtcSourceManager
|
||||
{
|
||||
private:
|
||||
srs_mutex_t lock;
|
||||
std::map<std::string, SrsRtcSource*> pool;
|
||||
public:
|
||||
SrsRtcSourceManager();
|
||||
virtual ~SrsRtcSourceManager();
|
||||
public:
|
||||
// create source when fetch from cache failed.
|
||||
// @param r the client request.
|
||||
// @param pps the matched source, if success never be NULL.
|
||||
virtual srs_error_t fetch_or_create(SrsRequest* r, SrsRtcSource** pps);
|
||||
private:
|
||||
// Get the exists source, NULL when not exists.
|
||||
// update the request and return the exists source.
|
||||
virtual SrsRtcSource* fetch(SrsRequest* r);
|
||||
};
|
||||
|
||||
// Global singleton instance.
|
||||
extern SrsRtcSourceManager* _srs_rtc_sources;
|
||||
|
||||
#endif
|
||||
|
Loading…
Reference in New Issue