iOS Stereo Playout/Recording

README.md

@@ -4,6 +4,10 @@ This repository is a fork of the WebRTC project. The original README can be foun
 
 ### Fork Specifics
 
+#### Features
+
+- Stereo playout/recording on iOS [#58](https://github.com/GetStream/webrtc/pull/58)
+
 #### `.gitignore`
 
 Due to the fork specifics, the repo's `.gitignore` has been updated to match the fork's requirements.

sdk/objc/native/api/audio_device_module.h

@@ -19,7 +19,8 @@
 namespace webrtc {
 
 // If `bypass_voice_processing` is true, WebRTC will attempt to disable hardware
-// audio processing on iOS.
+// audio processing on iOS. Stereo capture is automatically configured to bypass
+// voice processing even if this parameter is left as false.
 // Warning: Setting `bypass_voice_processing` will have unpredictable
 // consequences for the audio path in the device. It is not advisable to use in
 // most scenarios.

sdk/objc/native/src/audio/audio_device_ios.h

@@ -146,6 +146,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   int32_t SetStereoPlayout(bool enable) override;
   int32_t StereoPlayout(bool& enabled) const override;
   int32_t StereoRecordingIsAvailable(bool& available) override;
+  // Enabling stereo recording automatically bypasses the hardware voice
+  // processing (AEC/AGC) stage since the Voice Processing I/O unit only
+  // supports mono processing. See UpdateVoiceProcessingBypassRequirement().
   int32_t SetStereoRecording(bool enable) override;
   int32_t StereoRecording(bool& enabled) const override;
 
@@ -195,14 +198,38 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   // This method asks for the current hardware parameters and takes actions
   // if they should differ from what we have asked for initially. It also
   // defines `playout_parameters_` and `record_parameters_`.
-  void SetupAudioBuffersForActiveAudioSession();
+  void SetupAudioBuffersForActiveAudioSession() RTC_RUN_ON(thread_);
 
   // Creates the audio unit.
-  bool CreateAudioUnit();
+  bool CreateAudioUnit() RTC_RUN_ON(thread_);
 
   // Updates the audio unit state based on current state.
   void UpdateAudioUnit(bool can_play_or_record);
 
+  // Ensures the hardware voice-processing stage is bypassed when multi-channel
+  // playout or recording is requested. If the audio unit is already running,
+  // the updated state will be applied on the next initialization.
+  void UpdateVoiceProcessingBypassRequirement() RTC_RUN_ON(thread_);
+
+  // Updates the preferred channel counts stored in
+  // RTCAudioSessionConfiguration so forthcoming session activations request
+  // the desired layout.
+  void UpdateAudioSessionChannelPreferences() RTC_RUN_ON(thread_);
+
+  // Reconfigures the audio session and restarts the audio unit to apply
+  // updated channel counts while streaming. Returns true on success or when no
+  // reconfiguration is required.
+  bool ApplyChannelConfigurationChange() RTC_RUN_ON(thread_);
+
+  // Applies a channel configuration change and rolls back to the previous
+  // state if the update fails. Returns true when the change is successful.
+  bool ApplyChannelConfigurationChangeOrRollback(
+      const char* failure_log_message,
+      size_t previous_desired_playout,
+      size_t previous_desired_record,
+      const AudioParameters& previous_playout_parameters,
+      const AudioParameters& previous_record_parameters) RTC_RUN_ON(thread_);
+
   // Configures the audio session for WebRTC.
   bool ConfigureAudioSession();
 
@@ -223,7 +250,10 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   void PrepareForNewStart();
 
   // Determines whether voice processing should be enabled or disabled.
-  const bool bypass_voice_processing_;
+  const bool default_bypass_voice_processing_;
+  bool bypass_voice_processing_ RTC_GUARDED_BY(thread_);
+  size_t desired_playout_channels_ RTC_GUARDED_BY(thread_);
+  size_t desired_record_channels_ RTC_GUARDED_BY(thread_);
 
   // Handle a user speaking during muted event
   AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler_;