Audio Device Optimization (#29)

* Audio Device Optimization

allow listen-only mode in AudioUnit, adjust when category changes (#2)

release mic when category changes (#5)

Change defaults to iOS defaults (#7)

Sync audio session config (#8)

feat: support bypass voice processing for iOS. (#15)

Remove MacBookPro audio pan right code (#22)

fix: Fix can't open mic alone when built-in AEC is enabled. (#29)

feat: add audio device changes detect for windows. (#41)

fix Linux compile (#47)

AudioUnit: Don't rely on category switch for mic indicator to turn off (#52)

Stop recording on mute (turn off mic indicator) (#55)

Cherry pick audio selection from m97 release (#35)

[Mac] Allow audio device selection (#21)

RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80)

Co-authored-by: Hiroshi Horie <[email protected]>
Co-authored-by: David Zhao <[email protected]>

* fix compilation errors

---------

Co-authored-by: CloudWebRTC <[email protected]>
Co-authored-by: Hiroshi Horie <[email protected]>
Co-authored-by: David Zhao <[email protected]>

Author: 4 people

audio/audio_send_stream.cc

@@ -400,6 +400,11 @@ 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);
 }

audio/audio_send_stream.h

@@ -88,6 +88,7 @@ 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(

audio/audio_state.cc

@@ -149,8 +149,6 @@ void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
 #endif
         adm->StartRecording();
       }
-    } else {
-      RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
     }
   }
 }
@@ -160,7 +158,8 @@ 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();
   }
 }
@@ -216,6 +215,25 @@ void AudioState::UpdateNullAudioPollerState() {
     null_audio_poller_.Stop();
   }
 }
+
+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
 
 scoped_refptr<AudioState> AudioState::Create(const AudioState::Config& config) {

audio/audio_state.h

@@ -64,6 +64,9 @@ 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_{SequenceChecker::kDetached};
   const webrtc::AudioState::Config config_;

audio/channel_send.cc

@@ -163,6 +163,8 @@ class ChannelSend : public ChannelSendInterface,
   // Muting, Volume and Level.
   void SetInputMute(bool enable) override;
 
+  bool InputMute() const override;
+
   // Stats.
   ANAStats GetANAStatistics() const override;
 
@@ -235,8 +237,6 @@ class ChannelSend : public ChannelSendInterface,
                    size_t payloadSize,
                    int64_t absolute_capture_timestamp_ms) override;
 
-  bool InputMute() const;
-
   int32_t SendRtpAudio(AudioFrameType frameType,
                        uint8_t payloadType,
                        uint32_t rtp_timestamp_without_offset,

audio/channel_send.h

@@ -89,6 +89,8 @@ 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(

call/audio_send_stream.h

@@ -193,6 +193,7 @@ 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;