Do not initialize parameters with audio session if audio disabled

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

@@ -149,18 +149,6 @@ static void LogDeviceInfo() {
 #if !defined(NDEBUG)
  LogDeviceInfo();
 #endif
- // Store the preferred sample rate and preferred number of channels already
- // here. They have not been set and confirmed yet since configureForWebRTC
- // is not called until audio is about to start. However, it makes sense to
- // store the parameters now and then verify at a later stage.
- RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
- playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
- record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
- // Ensure that the audio device buffer (ADB) knows about the internal audio
- // parameters. Note that, even if we are unable to get a mono audio session,
- // we will always tell the I/O audio unit to do a channel format conversion
- // to guarantee mono on the "input side" of the audio unit.
- UpdateAudioDeviceBuffer();
  initialized_ = true;
  return InitStatus::OK;
 }
@@ -727,15 +715,17 @@ static void LogDeviceInfo() {
  // AttachAudioBuffer() is called at construction by the main class but check
  // just in case.
  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
- RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0);
- RTC_DCHECK_GT(record_parameters_.sample_rate(), 0);
- RTC_DCHECK_EQ(playout_parameters_.channels(), 1);
- RTC_DCHECK_EQ(record_parameters_.channels(), 1);
  // Inform the audio device buffer (ADB) about the new audio format.
- audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
- audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
- audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
- audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
+ if (playout_parameters_.is_valid()) {
+ RTC_DCHECK_EQ(playout_parameters_.channels(), 1);
+ audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
+ audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
+ }
+ if (record_parameters_.is_valid()) {
+ RTC_DCHECK_EQ(record_parameters_.channels(), 1);
+ audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
+ audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
+ }
 }
 
 void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
@@ -771,9 +761,9 @@ static void LogDeviceInfo() {
  // number of audio frames.
  // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
  // Hence, 128 is the size we expect to see in upcoming render callbacks.
- playout_parameters_.reset(sample_rate, playout_parameters_.channels(), io_buffer_duration);
+ playout_parameters_.reset(sample_rate, webRTCConfig.outputNumberOfChannels, io_buffer_duration);
  RTC_DCHECK(playout_parameters_.is_complete());
- record_parameters_.reset(sample_rate, record_parameters_.channels(), io_buffer_duration);
+ record_parameters_.reset(sample_rate, webRTCConfig.inputNumberOfChannels, io_buffer_duration);
  RTC_DCHECK(record_parameters_.is_complete());
  RTC_LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
  RTC_LOG(LS_INFO) << " bytes per I/O buffer: " << playout_parameters_.GetBytesPerBuffer();
@@ -943,6 +933,19 @@ static void LogDeviceInfo() {
  // If we are ready to play or record, and if the audio session can be
  // configured, then initialize the audio unit.
  if (session.canPlayOrRecord) {
+ // Store the preferred sample rate and preferred number of channels already
+ // here. They have not been set and confirmed yet since configureForWebRTC
+ // is not called until audio is about to start. However, it makes sense to
+ // store the parameters now and then verify at a later stage.
+ RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
+ playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
+ record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
+ // Ensure that the audio device buffer (ADB) knows about the internal audio
+ // parameters. Note that, even if we are unable to get a mono audio session,
+ // we will always tell the I/O audio unit to do a channel format conversion
+ // to guarantee mono on the "input side" of the audio unit.
+ UpdateAudioDeviceBuffer();
+
  // There should be no audio unit at this point.
  if (!CreateAudioUnit()) {
  [session unlockForConfiguration];