diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 7d6ec794d4..01e0ea8597 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -430,6 +430,11 @@ void AudioSendStream::SetMuted(bool muted) { channel_send_->SetInputMute(muted); } +bool AudioSendStream::GetMuted() { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + return channel_send_->InputMute(); +} + webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { return GetStats(true); } diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index 42be43afb9..03e2f05062 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -97,6 +97,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, int payload_frequency, int event, int duration_ms) override; + bool GetMuted() override; void SetMuted(bool muted) override; webrtc::AudioSendStream::Stats GetStats() const override; webrtc::AudioSendStream::Stats GetStats( diff --git a/audio/audio_state.cc b/audio/audio_state.cc index 7d2fb59956..673dd0651b 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc @@ -99,22 +99,26 @@ void AudioState::AddSendingStream(webrtc::AudioSendStream* stream, UpdateAudioTransportWithSendingStreams(); // Make sure recording is initialized; start recording if enabled. - auto* adm = config_.audio_device_module.get(); - if (!adm->Recording()) { - if (adm->InitRecording() == 0) { - if (recording_enabled_) { + if (ShouldRecord()) { + auto* adm = config_.audio_device_module.get(); + if (!adm->Recording()) { + if (adm->InitRecording() == 0) { + if (recording_enabled_) { + + // TODO: Verify if the following windows only logic is still required. #if defined(WEBRTC_WIN) - if (adm->BuiltInAECIsAvailable() && !adm->Playing()) { - if (!adm->PlayoutIsInitialized()) { - adm->InitPlayout(); + if (adm->BuiltInAECIsAvailable() && !adm->Playing()) { + if (!adm->PlayoutIsInitialized()) { + adm->InitPlayout(); + } + adm->StartPlayout(); } - adm->StartPlayout(); - } #endif - adm->StartRecording(); + adm->StartRecording(); + } + } else { + RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording."; } - } else { - RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording."; } } } @@ -124,7 +128,8 @@ void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) { auto count = sending_streams_.erase(stream); RTC_DCHECK_EQ(1, count); UpdateAudioTransportWithSendingStreams(); - if (sending_streams_.empty()) { + + if (!ShouldRecord()) { config_.audio_device_module->StopRecording(); } } @@ -152,7 +157,7 @@ void AudioState::SetRecording(bool enabled) { if (recording_enabled_ != enabled) { recording_enabled_ = enabled; if (enabled) { - if (!sending_streams_.empty()) { + if (ShouldRecord()) { config_.audio_device_module->StartRecording(); } } else { @@ -212,6 +217,39 @@ void AudioState::UpdateNullAudioPollerState() { null_audio_poller_.Stop(); } } + +void AudioState::OnMuteStreamChanged() { + + auto* adm = config_.audio_device_module.get(); + bool should_record = ShouldRecord(); + + if (should_record && !adm->Recording()) { + if (adm->InitRecording() == 0) { + adm->StartRecording(); + } + } else if (!should_record && adm->Recording()) { + adm->StopRecording(); + } +} + +bool AudioState::ShouldRecord() { + // no streams to send + if (sending_streams_.empty()) { + return false; + } + + int stream_count = sending_streams_.size(); + + int muted_count = 0; + for (const auto& kv : sending_streams_) { + if (kv.first->GetMuted()) { + muted_count++; + } + } + + return muted_count != stream_count; +} + } // namespace internal rtc::scoped_refptr AudioState::Create( diff --git a/audio/audio_state.h b/audio/audio_state.h index 6c2b7aa453..1d0d3be28a 100644 --- a/audio/audio_state.h +++ b/audio/audio_state.h @@ -47,6 +47,8 @@ class AudioState : public webrtc::AudioState { void SetStereoChannelSwapping(bool enable) override; + void OnMuteStreamChanged() override; + AudioDeviceModule* audio_device_module() { RTC_DCHECK(config_.audio_device_module); return config_.audio_device_module.get(); @@ -64,6 +66,9 @@ class AudioState : public webrtc::AudioState { void UpdateAudioTransportWithSendingStreams(); void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_); + // Returns true when at least 1 stream exists and all streams are not muted. + bool ShouldRecord(); + SequenceChecker thread_checker_; SequenceChecker process_thread_checker_; const webrtc::AudioState::Config config_; diff --git a/audio/channel_send.cc b/audio/channel_send.cc index fdc11a83fb..7248b120a7 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -98,6 +98,8 @@ class ChannelSend : public ChannelSendInterface, // Muting, Volume and Level. void SetInputMute(bool enable) override; + bool InputMute() const override; + // Stats. ANAStats GetANAStatistics() const override; @@ -163,6 +165,8 @@ class ChannelSend : public ChannelSendInterface, bool InputMute() const; + void OnUplinkPacketLossRate(float packet_loss_rate); + int32_t SendRtpAudio(AudioFrameType frameType, uint8_t payloadType, uint32_t rtp_timestamp, diff --git a/audio/channel_send.h b/audio/channel_send.h index cf9a273f70..d06e7fe5c4 100644 --- a/audio/channel_send.h +++ b/audio/channel_send.h @@ -95,6 +95,8 @@ class ChannelSendInterface { virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0; virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0; virtual int GetTargetBitrate() const = 0; + + virtual bool InputMute() const = 0; virtual void SetInputMute(bool muted) = 0; virtual void ProcessAndEncodeAudio( diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h index 0f42d0fb82..e6d8a3bc24 100644 --- a/call/audio_send_stream.h +++ b/call/audio_send_stream.h @@ -190,6 +190,7 @@ class AudioSendStream : public AudioSender { int event, int duration_ms) = 0; + virtual bool GetMuted() = 0; virtual void SetMuted(bool muted) = 0; virtual Stats GetStats() const = 0; diff --git a/call/audio_state.h b/call/audio_state.h index 79fb5cf981..85f04758dd 100644 --- a/call/audio_state.h +++ b/call/audio_state.h @@ -59,6 +59,9 @@ class AudioState : public rtc::RefCountInterface { virtual void SetStereoChannelSwapping(bool enable) = 0; + // Notify the AudioState that a stream updated it's mute state. + virtual void OnMuteStreamChanged() = 0; + static rtc::scoped_refptr Create( const AudioState::Config& config); diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index b6296a1993..b7c2e4c61c 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -2247,6 +2247,9 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) { ap->set_output_will_be_muted(all_muted); } + // Notfy the AudioState that the mute state has updated. + engine_->audio_state()->OnMuteStreamChanged(); + return true; } diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index daa964a655..616335a75b 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h @@ -87,6 +87,9 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { // Stops AEC dump. void StopAecDump() override; + // Moved to public so WebRtcVoiceMediaChannel can access it. + webrtc::AudioState* audio_state(); + private: // Every option that is "set" will be applied. Every option not "set" will be // ignored. This allows us to selectively turn on and off different options @@ -100,7 +103,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { webrtc::AudioDeviceModule* adm(); webrtc::AudioProcessing* apm() const; - webrtc::AudioState* audio_state(); std::vector CollectCodecs( const std::vector& specs) const;