webrtc-sdk · hiroshihorie · Oct 11, 2024 · Dec 5, 2024 · Dec 5, 2024 · Dec 10, 2024
diff --git a/audio/audio_send_stream.cc b/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);
 }

diff --git a/audio/audio_send_stream.h b/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(

diff --git a/audio/audio_state.cc b/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(

diff --git a/audio/audio_state.h b/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_;

diff --git a/audio/channel_send.cc b/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,

diff --git a/audio/channel_send.h b/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(

diff --git a/call/audio_send_stream.h b/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;

diff --git a/call/audio_state.h b/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);
 

diff --git a/media/engine/webrtc_voice_engine.cc b/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;
 }

diff --git a/media/engine/webrtc_voice_engine.h b/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;

diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc
@@ -307,8 +307,8 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
   }
 #endif  // WEBRTC_IOS
 
-  int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const override {
-    return impl_->SetAudioDeviceSink(sink);
+  int32_t SetObserver(AudioDeviceObserver* observer) const override {
+    return impl_->SetObserver(observer);
   }
 
  protected:

diff --git a/modules/audio_device/audio_device_generic.h b/modules/audio_device/audio_device_generic.h
@@ -135,7 +135,7 @@ class AudioDeviceGeneric {
   virtual int GetRecordAudioParameters(AudioParameters* params) const;
 #endif  // WEBRTC_IOS
 
-  virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) { return -1; }
+  virtual int32_t SetObserver(AudioDeviceObserver* observer) { return -1; }
   virtual int32_t GetPlayoutDevice() const { return -1; }
   virtual int32_t GetRecordingDevice() const { return -1; }
 

diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
@@ -34,11 +34,12 @@
 #if defined(WEBRTC_ENABLE_LINUX_PULSE)
 #include "modules/audio_device/linux/audio_device_pulse_linux.h"
 #endif
-#elif defined(WEBRTC_IOS)
-#include "sdk/objc/native/src/audio/audio_device_ios.h"
-#elif defined(WEBRTC_MAC)
-#include "modules/audio_device/mac/audio_device_mac.h"
 #endif
+
+#if defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
+#include "modules/audio_device/audio_engine_device.h"
+#endif
+
 #if defined(WEBRTC_DUMMY_FILE_DEVICES)
 #include "modules/audio_device/dummy/file_audio_device.h"
 #include "modules/audio_device/dummy/file_audio_device_factory.h"
@@ -246,16 +247,15 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
 // iOS ADM implementation.
 #if defined(WEBRTC_IOS)
   if (audio_layer == kPlatformDefaultAudio) {
-    audio_device_.reset(
-        new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/bypass_voice_processing_));
+    audio_device_.reset(new AudioEngineDevice(/*bypass_voice_processing=*/bypass_voice_processing_));
     RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
   }
 // END #if defined(WEBRTC_IOS)
 
 // Mac OS X ADM implementation.
 #elif defined(WEBRTC_MAC)
   if (audio_layer == kPlatformDefaultAudio) {
-    audio_device_.reset(new AudioDeviceMac());
+    audio_device_.reset(new AudioEngineDevice(/*bypass_voice_processing=*/false));
     RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized.";
   }
 #endif  // WEBRTC_MAC
@@ -902,9 +902,9 @@ int AudioDeviceModuleImpl::GetRecordAudioParameters(
 }
 #endif  // WEBRTC_IOS
 
-int32_t AudioDeviceModuleImpl::SetAudioDeviceSink(AudioDeviceSink* sink) const {
-  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << sink << ")";
-  int32_t ok = audio_device_->SetAudioDeviceSink(sink);
+int32_t AudioDeviceModuleImpl::SetObserver(AudioDeviceObserver* observer) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << observer << ")";
+  int32_t ok = audio_device_->SetObserver(observer);
   RTC_LOG(LS_INFO) << "output: " << ok;
   return ok;
 }

diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h
@@ -156,7 +156,7 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
   int GetRecordAudioParameters(AudioParameters* params) const override;
 #endif  // WEBRTC_IOS
 
-  int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const override;
+  int32_t SetObserver(AudioDeviceObserver* observer) const override;
   int32_t GetPlayoutDevice() const override;
   int32_t GetRecordingDevice() const override;