| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "audio/channel_proxy.h" |
| |
| #include <utility> |
| |
| #include "api/call/audio_sink.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| |
| namespace webrtc { |
| namespace voe { |
| ChannelProxy::ChannelProxy() {} |
| |
| ChannelProxy::ChannelProxy(std::unique_ptr<Channel> channel) |
| : channel_(std::move(channel)) { |
| RTC_DCHECK(channel_); |
| module_process_thread_checker_.DetachFromThread(); |
| } |
| |
| ChannelProxy::~ChannelProxy() {} |
| |
| bool ChannelProxy::SetEncoder(int payload_type, |
| std::unique_ptr<AudioEncoder> encoder) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->SetEncoder(payload_type, std::move(encoder)); |
| } |
| |
| void ChannelProxy::ModifyEncoder( |
| rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->ModifyEncoder(modifier); |
| } |
| |
| void ChannelProxy::SetRTCPStatus(bool enable) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetRTCPStatus(enable); |
| } |
| |
| void ChannelProxy::SetLocalSSRC(uint32_t ssrc) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| int error = channel_->SetLocalSSRC(ssrc); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| void ChannelProxy::SetMid(const std::string& mid, int extension_id) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetMid(mid, extension_id); |
| } |
| |
| void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array. |
| std::string c_name_limited = c_name.substr(0, 255); |
| int error = channel_->SetRTCP_CNAME(c_name_limited.c_str()); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| void ChannelProxy::SetNACKStatus(bool enable, int max_packets) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetNACKStatus(enable, max_packets); |
| } |
| |
| void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| int error = channel_->SetSendAudioLevelIndicationStatus(enable, id); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| void ChannelProxy::EnableSendTransportSequenceNumber(int id) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->EnableSendTransportSequenceNumber(id); |
| } |
| |
| void ChannelProxy::RegisterSenderCongestionControlObjects( |
| RtpTransportControllerSendInterface* transport, |
| RtcpBandwidthObserver* bandwidth_observer) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->RegisterSenderCongestionControlObjects(transport, |
| bandwidth_observer); |
| } |
| |
| void ChannelProxy::RegisterReceiverCongestionControlObjects( |
| PacketRouter* packet_router) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->RegisterReceiverCongestionControlObjects(packet_router); |
| } |
| |
| void ChannelProxy::ResetSenderCongestionControlObjects() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->ResetSenderCongestionControlObjects(); |
| } |
| |
| void ChannelProxy::ResetReceiverCongestionControlObjects() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->ResetReceiverCongestionControlObjects(); |
| } |
| |
| CallStatistics ChannelProxy::GetRTCPStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| CallStatistics stats = {0}; |
| int error = channel_->GetRTPStatistics(stats); |
| RTC_DCHECK_EQ(0, error); |
| return stats; |
| } |
| |
| std::vector<ReportBlock> ChannelProxy::GetRemoteRTCPReportBlocks() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| std::vector<webrtc::ReportBlock> blocks; |
| int error = channel_->GetRemoteRTCPReportBlocks(&blocks); |
| RTC_DCHECK_EQ(0, error); |
| return blocks; |
| } |
| |
| NetworkStatistics ChannelProxy::GetNetworkStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| NetworkStatistics stats = {0}; |
| int error = channel_->GetNetworkStatistics(stats); |
| RTC_DCHECK_EQ(0, error); |
| return stats; |
| } |
| |
| AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| AudioDecodingCallStats stats; |
| channel_->GetDecodingCallStatistics(&stats); |
| return stats; |
| } |
| |
| ANAStats ChannelProxy::GetANAStatistics() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetANAStatistics(); |
| } |
| |
| int ChannelProxy::GetSpeechOutputLevelFullRange() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetSpeechOutputLevelFullRange(); |
| } |
| |
| double ChannelProxy::GetTotalOutputEnergy() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetTotalOutputEnergy(); |
| } |
| |
| double ChannelProxy::GetTotalOutputDuration() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetTotalOutputDuration(); |
| } |
| |
| uint32_t ChannelProxy::GetDelayEstimate() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() || |
| module_process_thread_checker_.CalledOnValidThread()); |
| return channel_->GetDelayEstimate(); |
| } |
| |
| bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type, |
| int payload_frequency) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->SetSendTelephoneEventPayloadType(payload_type, |
| payload_frequency) == 0; |
| } |
| |
| bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->SendTelephoneEventOutband(event, duration_ms) == 0; |
| } |
| |
| void ChannelProxy::SetBitrate(int bitrate_bps, int64_t probing_interval_ms) { |
| // This method can be called on the worker thread, module process thread |
| // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged. |
| // TODO(solenberg): Figure out a good way to check this or enforce calling |
| // rules. |
| // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() || |
| // module_process_thread_checker_.CalledOnValidThread()); |
| channel_->SetBitRate(bitrate_bps, probing_interval_ms); |
| } |
| |
| void ChannelProxy::SetReceiveCodecs( |
| const std::map<int, SdpAudioFormat>& codecs) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetReceiveCodecs(codecs); |
| } |
| |
| void ChannelProxy::SetSink(AudioSinkInterface* sink) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetSink(sink); |
| } |
| |
| void ChannelProxy::SetInputMute(bool muted) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetInputMute(muted); |
| } |
| |
| void ChannelProxy::RegisterTransport(Transport* transport) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->RegisterTransport(transport); |
| } |
| |
| void ChannelProxy::OnRtpPacket(const RtpPacketReceived& packet) { |
| // May be called on either worker thread or network thread. |
| channel_->OnRtpPacket(packet); |
| } |
| |
| bool ChannelProxy::ReceivedRTCPPacket(const uint8_t* packet, size_t length) { |
| // May be called on either worker thread or network thread. |
| return channel_->ReceivedRTCPPacket(packet, length) == 0; |
| } |
| |
| void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetChannelOutputVolumeScaling(scaling); |
| } |
| |
| AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo( |
| int sample_rate_hz, |
| AudioFrame* audio_frame) { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| return channel_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); |
| } |
| |
| int ChannelProxy::PreferredSampleRate() const { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| return channel_->PreferredSampleRate(); |
| } |
| |
| void ChannelProxy::ProcessAndEncodeAudio( |
| std::unique_ptr<AudioFrame> audio_frame) { |
| RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_); |
| return channel_->ProcessAndEncodeAudio(std::move(audio_frame)); |
| } |
| |
| void ChannelProxy::SetTransportOverhead(int transport_overhead_per_packet) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetTransportOverhead(transport_overhead_per_packet); |
| } |
| |
| void ChannelProxy::AssociateSendChannel( |
| const ChannelProxy& send_channel_proxy) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetAssociatedSendChannel(send_channel_proxy.channel_.get()); |
| } |
| |
| void ChannelProxy::DisassociateSendChannel() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->SetAssociatedSendChannel(nullptr); |
| } |
| |
| RtpRtcp* ChannelProxy::GetRtpRtcp() const { |
| RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread()); |
| return channel_->GetRtpRtcp(); |
| } |
| |
| absl::optional<Syncable::Info> ChannelProxy::GetSyncInfo() const { |
| RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread()); |
| return channel_->GetSyncInfo(); |
| } |
| |
| uint32_t ChannelProxy::GetPlayoutTimestamp() const { |
| RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_); |
| unsigned int timestamp = 0; |
| int error = channel_->GetPlayoutTimestamp(timestamp); |
| RTC_DCHECK(!error || timestamp == 0); |
| return timestamp; |
| } |
| |
| void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) { |
| RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread()); |
| // Limit to range accepted by both VoE and ACM, so we're at least getting as |
| // close as possible, instead of failing. |
| delay_ms = rtc::SafeClamp(delay_ms, 0, 10000); |
| int error = channel_->SetMinimumPlayoutDelay(delay_ms); |
| if (0 != error) { |
| RTC_LOG(LS_WARNING) << "Error setting minimum playout delay."; |
| } |
| } |
| |
| bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetRecCodec(*codec_inst) == 0; |
| } |
| |
| void ChannelProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->OnTwccBasedUplinkPacketLossRate(packet_loss_rate); |
| } |
| |
| void ChannelProxy::OnRecoverableUplinkPacketLossRate( |
| float recoverable_packet_loss_rate) { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate); |
| } |
| |
| std::vector<RtpSource> ChannelProxy::GetSources() const { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| return channel_->GetSources(); |
| } |
| |
| void ChannelProxy::StartSend() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| int error = channel_->StartSend(); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| void ChannelProxy::StopSend() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| channel_->StopSend(); |
| } |
| |
| void ChannelProxy::StartPlayout() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| int error = channel_->StartPlayout(); |
| RTC_DCHECK_EQ(0, error); |
| } |
| |
| void ChannelProxy::StopPlayout() { |
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
| int error = channel_->StopPlayout(); |
| RTC_DCHECK_EQ(0, error); |
| } |
| } // namespace voe |
| } // namespace webrtc |