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 <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io> * fix compilation errors --------- Co-authored-by: CloudWebRTC <duanweiwei1982@gmail.com> Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io>

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;