Squashed commit of the following:

commit 98dc0ac
Author: Hiroshi Horie <[email protected]>
Date:   Thu Jan 23 03:27:20 2025 +0900

    Rendering fix

commit 345f8b7
Author: Hiroshi Horie <[email protected]>
Date:   Thu Jan 23 00:41:35 2025 +0900

    Manual rendering

commit be003d5
Author: Hiroshi Horie <[email protected]>
Date:   Wed Jan 22 12:03:18 2025 +0900

    RTCAudioDeviceModuleDelegate

commit 2babb14
Author: Hiroshi Horie <[email protected]>
Date:   Fri Jan 17 16:18:39 2025 +0900

    Squashed recent improvements

    Pre initialize mode
    Pre initialize logic
    Persistent
    Checks
    Fix buffer logic
    Patch default input_mute state
    Buffer checks
    Start buffer on enable
    Delay estimate 0
    Stop engine on interrupt
    Pass should_resume
    Silence warning
    Correct session config
    Fix state
    Start logic
    Misc
    Rem ses
    Rem ses2
    State helper
    Minor patch
    Simplify
    Change stop create order
    Working state
    Ref
    State helpers

commit 235da97
Author: Hiroshi Horie <[email protected]>
Date:   Tue Jan 14 15:52:32 2025 +0900

    Squashed recent progress

commit 6ba820c
Author: Hiroshi Horie <[email protected]>
Date:   Tue Dec 31 01:57:09 2024 +0900

    Squashed recent progress

    Fix adm selection

    Fixes

    Revert adm selection in audio_device_impl

    Rename IsManualRenderingMode

    Simplify pcm buffer delegate

    Fixes

    Fixes

    Ducking config

    Strip manual rendering logic

    Runtime-ducking config

    Fix compile

    Fix start recording

    Connect output

    Buffer logic

    Enable output when input is enabled

commit 49ca1ee
Author: Hiroshi Horie <[email protected]>
Date:   Sun Dec 22 04:32:07 2024 +0700

    Check AGC

commit 5bbeb48
Author: Hiroshi Horie <[email protected]>
Date:   Sat Dec 21 14:15:31 2024 +0700

    Debug print audio graph

commit 631126f
Author: Hiroshi Horie <[email protected]>
Date:   Fri Dec 20 23:24:42 2024 +0700

    Fix macOS vp

commit e07b814
Author: Hiroshi Horie <[email protected]>
Date:   Tue Dec 17 12:28:55 2024 +0700

    Clean up imports

commit 1bdb158
Author: Hiroshi Horie <[email protected]>
Date:   Wed Dec 11 12:35:12 2024 +0700

    Muted talker detection

commit 0324b22
Author: Hiroshi Horie <[email protected]>
Date:   Mon Dec 16 21:11:45 2024 +0700

    Rename AudioDeviceSink

commit ed22ffb
Author: Hiroshi Horie <[email protected]>
Date:   Tue Dec 17 01:11:23 2024 +0700

    Move to private method

commit db00fe4
Author: Hiroshi Horie <[email protected]>
Date:   Wed Dec 11 00:16:55 2024 +0700

    Other audio ducking

commit a7282bd
Author: Hiroshi Horie <[email protected]>
Date:   Thu Dec 5 15:27:34 2024 +0700

    AudioEngine

commit d31187b
Author: Hiroshi Horie <[email protected]>
Date:   Thu Dec 5 15:26:38 2024 +0700

    Connect voice engine mute to adm

commit 3df68d4
Author: Hiroshi Horie <[email protected]>
Date:   Fri Oct 11 20:41:44 2024 +0900

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

    This reverts commit c0209ef.

Author: hiroshihorie

audio/audio_send_stream.cc

@@ -415,11 +415,6 @@ 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

@@ -94,7 +94,6 @@ 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

@@ -98,26 +98,22 @@ void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
   UpdateAudioTransportWithSendingStreams();
 
   // Make sure recording is initialized; start recording if 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.
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Recording()) {
+    if (adm->InitRecording() == 0) {
+      if (recording_enabled_) {
 #if defined(WEBRTC_WIN)
-          if (adm->BuiltInAECIsAvailable() && !adm->Playing()) {
-            if (!adm->PlayoutIsInitialized()) {
-              adm->InitPlayout();
-            }
-            adm->StartPlayout();
+        if (adm->BuiltInAECIsAvailable() && !adm->Playing()) {
+          if (!adm->PlayoutIsInitialized()) {
+            adm->InitPlayout();
           }
-#endif
-          adm->StartRecording();
+          adm->StartPlayout();
         }
-      } else {
-        RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
+#endif
+        adm->StartRecording();
       }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
     }
   }
 }
@@ -127,10 +123,7 @@ void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) {
   auto count = sending_streams_.erase(stream);
   RTC_DCHECK_EQ(1, count);
   UpdateAudioTransportWithSendingStreams();
-
-  bool should_record = ShouldRecord();
-  RTC_LOG(LS_INFO) << "RemoveSendingStream: should_record = " << should_record;
-  if (!should_record) {
+  if (sending_streams_.empty()) {
     config_.audio_device_module->StopRecording();
   }
 }
@@ -158,7 +151,7 @@ void AudioState::SetRecording(bool enabled) {
   if (recording_enabled_ != enabled) {
     recording_enabled_ = enabled;
     if (enabled) {
-      if (ShouldRecord()) {
+      if (!sending_streams_.empty()) {
         config_.audio_device_module->StartRecording();
       }
     } else {
@@ -218,43 +211,6 @@ void AudioState::UpdateNullAudioPollerState() {
     null_audio_poller_.Stop();
   }
 }
-
-void AudioState::OnMuteStreamChanged() {
-
-  auto* adm = config_.audio_device_module.get();
-  bool should_record = ShouldRecord();
-
-  RTC_LOG(LS_INFO) << "OnMuteStreamChanged: should_record = " << should_record;
-  if (should_record && !adm->Recording()) {
-    if (adm->InitRecording() == 0) {
-      adm->StartRecording();
-    }
-  } else if (!should_record && adm->Recording()) {
-    adm->StopRecording();
-  }
-}
-
-bool AudioState::ShouldRecord() {
-  RTC_LOG(LS_INFO) << "ShouldRecord";
-  // no streams to send
-  if (sending_streams_.empty()) {
-    RTC_LOG(LS_INFO) << "ShouldRecord: send stream = 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++;
-    }
-  }
-
-  RTC_LOG(LS_INFO) << "ShouldRecord: " << muted_count << " muted, " << stream_count << " sending";
-  return muted_count != stream_count;
-}
-
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(

audio/audio_state.h

@@ -47,8 +47,6 @@ 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();
@@ -66,9 +64,6 @@ 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

@@ -100,8 +100,6 @@ class ChannelSend : public ChannelSendInterface,
   // Muting, Volume and Level.
   void SetInputMute(bool enable) override;
 
-  bool InputMute() const override;
-
   // Stats.
   ANAStats GetANAStatistics() const override;
 
@@ -165,6 +163,8 @@ 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

@@ -83,8 +83,6 @@ 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

@@ -190,7 +190,6 @@ 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;

call/audio_state.h

@@ -59,9 +59,6 @@ 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<AudioState> Create(
       const AudioState::Config& config);
 

media/engine/webrtc_voice_engine.cc

@@ -1698,8 +1698,10 @@ bool WebRtcVoiceSendChannel::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();
+  webrtc::AudioDeviceModule* adm = engine()->adm();
+  if (adm) {
+    adm->SetMicrophoneMute(all_muted);
+  }
 
   return true;
 }

media/engine/webrtc_voice_engine.h

@@ -132,8 +132,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
 
   absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats()
       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
@@ -147,6 +145,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
 
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm() const;
+  webrtc::AudioState* audio_state();
 
   std::vector<AudioCodec> CollectCodecs(
       const std::vector<webrtc::AudioCodecSpec>& specs) const;