
Should fix deadlock on build bots. Before, TraceImpl called TraceDispatcher::Print, while TraceDispatcher::Deregister called TraceImpl through VideoEngine::SetTraceCallback. This violates locking order as both take their own locks. BUG=2421 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2340005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4905 4adac7df-926f-26a2-2b94-8c16560cd09d
309 lines
8.8 KiB
C++
309 lines
8.8 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/internal/call.h"
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#include <assert.h>
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#include <string.h>
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#include <map>
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#include <vector>
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/video_engine/include/vie_base.h"
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#include "webrtc/video_engine/include/vie_codec.h"
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#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
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#include "webrtc/video_engine/internal/video_receive_stream.h"
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#include "webrtc/video_engine/internal/video_send_stream.h"
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namespace webrtc {
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class TraceDispatcher : public TraceCallback {
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public:
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TraceDispatcher()
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: crit_(CriticalSectionWrapper::CreateCriticalSection()),
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initialized_(false),
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filter_(kTraceNone) {}
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~TraceDispatcher() {
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if (initialized_) {
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Trace::ReturnTrace();
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VideoEngine::SetTraceCallback(NULL);
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}
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}
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virtual void Print(TraceLevel level,
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const char* message,
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int length) OVERRIDE {
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CriticalSectionScoped lock(crit_.get());
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for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
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it != callbacks_.end();
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++it) {
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if ((level & it->second->trace_filter) != kTraceNone)
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it->second->trace_callback->Print(level, message, length);
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}
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}
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void RegisterCallback(Call* call, Call::Config* config) {
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if (config->trace_callback == NULL)
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return;
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CriticalSectionScoped lock(crit_.get());
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callbacks_[call] = config;
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filter_ |= config->trace_filter;
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if (filter_ != kTraceNone && !initialized_) {
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initialized_ = true;
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Trace::CreateTrace();
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VideoEngine::SetTraceCallback(this);
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}
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VideoEngine::SetTraceFilter(filter_);
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}
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void DeregisterCallback(Call* call) {
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CriticalSectionScoped lock(crit_.get());
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callbacks_.erase(call);
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filter_ = kTraceNone;
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for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
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it != callbacks_.end();
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++it) {
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filter_ |= it->second->trace_filter;
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}
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VideoEngine::SetTraceFilter(filter_);
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}
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private:
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scoped_ptr<CriticalSectionWrapper> crit_;
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bool initialized_;
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unsigned int filter_;
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std::map<Call*, Call::Config*> callbacks_;
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};
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namespace internal {
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TraceDispatcher* global_trace_dispatcher = NULL;
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} // internal
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Call* Call::Create(const Call::Config& config) {
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if (internal::global_trace_dispatcher == NULL) {
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TraceDispatcher* dispatcher = new TraceDispatcher();
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// TODO(pbos): Atomic compare and exchange.
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if (internal::global_trace_dispatcher == NULL) {
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internal::global_trace_dispatcher = dispatcher;
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} else {
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delete dispatcher;
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}
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}
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VideoEngine* video_engine = VideoEngine::Create();
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assert(video_engine != NULL);
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return new internal::Call(video_engine, config);
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}
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namespace internal {
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Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
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: config_(config),
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receive_lock_(RWLockWrapper::CreateRWLock()),
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send_lock_(RWLockWrapper::CreateRWLock()),
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rtp_header_parser_(RtpHeaderParser::Create()),
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video_engine_(video_engine) {
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assert(video_engine != NULL);
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assert(config.send_transport != NULL);
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global_trace_dispatcher->RegisterCallback(this, &config_);
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rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
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assert(rtp_rtcp_ != NULL);
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codec_ = ViECodec::GetInterface(video_engine_);
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assert(codec_ != NULL);
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}
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Call::~Call() {
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global_trace_dispatcher->DeregisterCallback(this);
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codec_->Release();
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rtp_rtcp_->Release();
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webrtc::VideoEngine::Delete(video_engine_);
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}
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PacketReceiver* Call::Receiver() { return this; }
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std::vector<VideoCodec> Call::GetVideoCodecs() {
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std::vector<VideoCodec> codecs;
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VideoCodec codec;
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for (size_t i = 0; i < static_cast<size_t>(codec_->NumberOfCodecs()); ++i) {
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if (codec_->GetCodec(i, codec) == 0) {
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codecs.push_back(codec);
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}
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}
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return codecs;
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}
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VideoSendStream::Config Call::GetDefaultSendConfig() {
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VideoSendStream::Config config;
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codec_->GetCodec(0, config.codec);
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return config;
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}
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VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
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assert(config.rtp.ssrcs.size() > 0);
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assert(config.codec.numberOfSimulcastStreams == 0 ||
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config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
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VideoSendStream* send_stream = new VideoSendStream(
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config_.send_transport, config_.overuse_detection, video_engine_, config);
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WriteLockScoped write_lock(*send_lock_);
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for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
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assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
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send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
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}
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return send_stream;
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}
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SendStreamState* Call::DestroySendStream(webrtc::VideoSendStream* send_stream) {
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assert(send_stream != NULL);
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VideoSendStream* send_stream_impl = NULL;
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{
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WriteLockScoped write_lock(*send_lock_);
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for (std::map<uint32_t, VideoSendStream*>::iterator it =
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send_ssrcs_.begin();
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it != send_ssrcs_.end();
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++it) {
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if (it->second == static_cast<VideoSendStream*>(send_stream)) {
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send_stream_impl = it->second;
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send_ssrcs_.erase(it);
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break;
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}
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}
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}
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assert(send_stream_impl != NULL);
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delete send_stream_impl;
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// TODO(pbos): Return its previous state
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return NULL;
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}
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VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
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return VideoReceiveStream::Config();
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}
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VideoReceiveStream* Call::CreateReceiveStream(
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const VideoReceiveStream::Config& config) {
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VideoReceiveStream* receive_stream =
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new VideoReceiveStream(video_engine_, config, config_.send_transport);
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WriteLockScoped write_lock(*receive_lock_);
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assert(receive_ssrcs_.find(config.rtp.ssrc) == receive_ssrcs_.end());
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receive_ssrcs_[config.rtp.ssrc] = receive_stream;
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return receive_stream;
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}
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void Call::DestroyReceiveStream(webrtc::VideoReceiveStream* receive_stream) {
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assert(receive_stream != NULL);
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VideoReceiveStream* receive_stream_impl = NULL;
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{
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WriteLockScoped write_lock(*receive_lock_);
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for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
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receive_ssrcs_.begin();
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it != receive_ssrcs_.end();
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++it) {
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if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
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receive_stream_impl = it->second;
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receive_ssrcs_.erase(it);
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break;
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}
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}
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}
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assert(receive_stream_impl != NULL);
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delete receive_stream_impl;
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}
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uint32_t Call::SendBitrateEstimate() {
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// TODO(pbos): Return send-bitrate estimate
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return 0;
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}
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uint32_t Call::ReceiveBitrateEstimate() {
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// TODO(pbos): Return receive-bitrate estimate
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return 0;
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}
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bool Call::DeliverRtcp(const uint8_t* packet, size_t length) {
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// TODO(pbos): Figure out what channel needs it actually.
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// Do NOT broadcast! Also make sure it's a valid packet.
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bool rtcp_delivered = false;
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{
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ReadLockScoped read_lock(*receive_lock_);
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for (std::map<uint32_t, VideoReceiveStream*>::iterator it =
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receive_ssrcs_.begin();
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it != receive_ssrcs_.end();
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++it) {
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if (it->second->DeliverRtcp(packet, length))
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rtcp_delivered = true;
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}
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}
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{
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ReadLockScoped read_lock(*send_lock_);
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for (std::map<uint32_t, VideoSendStream*>::iterator it =
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send_ssrcs_.begin();
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it != send_ssrcs_.end();
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++it) {
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if (it->second->DeliverRtcp(packet, length))
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rtcp_delivered = true;
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}
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}
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return rtcp_delivered;
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}
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bool Call::DeliverRtp(const RTPHeader& header,
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const uint8_t* packet,
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size_t length) {
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VideoReceiveStream* receiver;
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{
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ReadLockScoped read_lock(*receive_lock_);
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std::map<uint32_t, VideoReceiveStream*>::iterator it =
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receive_ssrcs_.find(header.ssrc);
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if (it == receive_ssrcs_.end()) {
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// TODO(pbos): Log some warning, SSRC without receiver.
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return false;
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}
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receiver = it->second;
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}
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return receiver->DeliverRtp(static_cast<const uint8_t*>(packet), length);
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}
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bool Call::DeliverPacket(const uint8_t* packet, size_t length) {
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// TODO(pbos): ExtensionMap if there are extensions.
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if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)))
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return DeliverRtcp(packet, length);
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RTPHeader rtp_header;
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if (!rtp_header_parser_->Parse(packet, static_cast<int>(length), &rtp_header))
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return false;
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return DeliverRtp(rtp_header, packet, length);
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}
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} // namespace internal
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} // namespace webrtc
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