Remove unused/unnecessary things from ChannelSend.
Bug: none Change-Id: I48e105d39597c3a84402599af7289f2ea9adc0c6 Reviewed-on: https://webrtc-review.googlesource.com/c/111183 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25671}
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@ -199,7 +199,6 @@ AudioSendStream::AudioSendStream(
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RTC_DCHECK(rtp_transport || config.media_transport);
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RTC_DCHECK(overall_call_lifetime_);
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channel_proxy_->SetRTCPStatus(true);
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rtp_rtcp_module_ = channel_proxy_->GetRtpRtcp();
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RTC_DCHECK(rtp_rtcp_module_);
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@ -193,7 +193,6 @@ struct ConfigHelper {
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EXPECT_CALL(*channel_proxy_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
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return &this->rtp_rtcp_;
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}));
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EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
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EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
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EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
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EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
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@ -442,12 +442,6 @@ bool ChannelSend::SendRtcp(const uint8_t* data, size_t len) {
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return true;
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}
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int ChannelSend::PreferredSampleRate() const {
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// Return the bigger of playout and receive frequency in the ACM.
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return std::max(audio_coding_->ReceiveFrequency(),
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audio_coding_->PlayoutFrequency());
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}
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ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
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ProcessThread* module_process_thread,
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MediaTransportInterface* media_transport,
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@ -505,26 +499,14 @@ ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
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_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
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_rtpRtcpModule->SetSendingMediaStatus(false);
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Init();
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}
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ChannelSend::~ChannelSend() {
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Terminate();
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RTC_DCHECK(!channel_state_.Get().sending);
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}
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void ChannelSend::Init() {
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channel_state_.Reset();
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// --- Add modules to process thread (for periodic schedulation)
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_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
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// --- ACM initialization
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int error = audio_coding_->InitializeReceiver();
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RTC_DCHECK_EQ(0, error);
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// --- RTP/RTCP module initialization
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// Ensure that RTCP is enabled by default for the created channel.
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// Note that, the module will keep generating RTCP until it is explicitly
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// disabled by the user.
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@ -533,29 +515,22 @@ void ChannelSend::Init() {
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// RTCP is enabled by default.
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_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
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// --- Register all permanent callbacks
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error = audio_coding_->RegisterTransportCallback(this);
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RTC_DCHECK_EQ(0, error);
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}
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void ChannelSend::Terminate() {
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ChannelSend::~ChannelSend() {
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RTC_DCHECK(construction_thread_.CalledOnValidThread());
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// Must be called on the same thread as Init().
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StopSend();
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// The order to safely shutdown modules in a channel is:
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// 1. De-register callbacks in modules
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// 2. De-register modules in process thread
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// 3. Destroy modules
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int error = audio_coding_->RegisterTransportCallback(NULL);
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RTC_DCHECK_EQ(0, error);
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// De-register modules in process thread
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if (_moduleProcessThreadPtr)
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_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
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// End of modules shutdown
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RTC_DCHECK(!channel_state_.Get().sending);
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}
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void ChannelSend::StartSend() {
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@ -853,10 +828,6 @@ void ChannelSend::ResetSenderCongestionControlObjects() {
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rtp_packet_sender_proxy_->SetPacketSender(nullptr);
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}
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void ChannelSend::SetRTCPStatus(bool enable) {
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_rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
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}
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void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
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// Note: SetCNAME() accepts a c string of length at most 255.
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const std::string c_name_limited(c_name.substr(0, 255));
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@ -175,7 +175,6 @@ class ChannelSend
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RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer);
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void ResetSenderCongestionControlObjects();
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void SetRTCPStatus(bool enable);
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void SetRTCP_CNAME(absl::string_view c_name);
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std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
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CallSendStatistics GetRTCPStatistics() const;
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@ -212,9 +211,6 @@ class ChannelSend
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private:
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class ProcessAndEncodeAudioTask;
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void Init();
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void Terminate();
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// From AudioPacketizationCallback in the ACM
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int32_t SendData(FrameType frameType,
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uint8_t payloadType,
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@ -229,8 +225,6 @@ class ChannelSend
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const PacketOptions& packet_options) override;
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bool SendRtcp(const uint8_t* data, size_t len) override;
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int PreferredSampleRate() const;
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bool Sending() const { return channel_state_.Get().sending; }
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// From OverheadObserver in the RTP/RTCP module
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@ -66,11 +66,6 @@ void ChannelSendProxy::ModifyEncoder(
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channel_->ModifyEncoder(modifier);
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}
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void ChannelSendProxy::SetRTCPStatus(bool enable) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetRTCPStatus(enable);
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}
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void ChannelSendProxy::SetMid(const std::string& mid, int extension_id) {
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RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
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channel_->SetMid(mid, extension_id);
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@ -55,7 +55,6 @@ class ChannelSendProxy {
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virtual void ModifyEncoder(
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rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
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virtual void SetRTCPStatus(bool enable);
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virtual void SetMid(const std::string& mid, int extension_id);
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virtual void SetRTCP_CNAME(absl::string_view c_name);
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virtual void SetExtmapAllowMixed(bool extmap_allow_mixed);
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@ -72,7 +72,6 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
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MOCK_METHOD1(
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ModifyEncoder,
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void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
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MOCK_METHOD1(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name));
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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