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

* initial impl * more comments * more comment * adjust indent * comments
webrtc-sdk · Jan 18, 2023 · 8670af4 · 8670af4
1 parent f125bf4
commit 8670af4
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Showing 10 changed files with 79 additions and 15 deletions.
diff --git 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
@@ -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
@@ -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> AudioState::Create(

diff --git 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
@@ -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
@@ -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
@@ -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
@@ -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<AudioState> Create(
       const AudioState::Config& config);
 

diff --git 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
@@ -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<AudioCodec> CollectCodecs(
       const std::vector<webrtc::AudioCodecSpec>& specs) const;