diff options
Diffstat (limited to 'third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm')
| -rw-r--r-- | third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm | 711 |
1 files changed, 711 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm new file mode 100644 index 0000000000..d629fae20f --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm @@ -0,0 +1,711 @@ +/* + * Copyright (c) 2022 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "objc_audio_device.h" +#include "objc_audio_device_delegate.h" + +#import "components/audio/RTCAudioDevice.h" +#include "modules/audio_device/fine_audio_buffer.h" + +#include "api/task_queue/default_task_queue_factory.h" +#include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/time_utils.h" + +namespace { + +webrtc::AudioParameters RecordParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) { + const double sample_rate = static_cast<int>([audio_device deviceInputSampleRate]); + const size_t channels = static_cast<size_t>([audio_device inputNumberOfChannels]); + const size_t frames_per_buffer = + static_cast<size_t>(sample_rate * [audio_device inputIOBufferDuration] + .5); + return webrtc::AudioParameters(sample_rate, channels, frames_per_buffer); +} + +webrtc::AudioParameters PlayoutParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) { + const double sample_rate = static_cast<int>([audio_device deviceOutputSampleRate]); + const size_t channels = static_cast<size_t>([audio_device outputNumberOfChannels]); + const size_t frames_per_buffer = + static_cast<size_t>(sample_rate * [audio_device outputIOBufferDuration] + .5); + return webrtc::AudioParameters(sample_rate, channels, frames_per_buffer); +} + +} // namespace + +namespace webrtc { +namespace objc_adm { + +ObjCAudioDeviceModule::ObjCAudioDeviceModule(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) + : audio_device_(audio_device), task_queue_factory_(CreateDefaultTaskQueueFactory()) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK(audio_device_); + thread_checker_.Detach(); + io_playout_thread_checker_.Detach(); + io_record_thread_checker_.Detach(); +} + +ObjCAudioDeviceModule::~ObjCAudioDeviceModule() { + RTC_DLOG_F(LS_VERBOSE) << ""; +} + +int32_t ObjCAudioDeviceModule::RegisterAudioCallback(AudioTransport* audioCallback) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK(audio_device_buffer_); + return audio_device_buffer_->RegisterAudioCallback(audioCallback); +} + +int32_t ObjCAudioDeviceModule::Init() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (Initialized()) { + RTC_LOG_F(LS_INFO) << "Already initialized"; + return 0; + } + io_playout_thread_checker_.Detach(); + io_record_thread_checker_.Detach(); + + thread_ = rtc::Thread::Current(); + audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer(task_queue_factory_.get())); + + if (![audio_device_ isInitialized]) { + if (audio_device_delegate_ == nil) { + audio_device_delegate_ = [[ObjCAudioDeviceDelegate alloc] + initWithAudioDeviceModule:rtc::scoped_refptr<ObjCAudioDeviceModule>(this) + audioDeviceThread:thread_]; + } + + if (![audio_device_ initializeWithDelegate:audio_device_delegate_]) { + RTC_LOG_F(LS_WARNING) << "Failed to initialize audio device"; + [audio_device_delegate_ resetAudioDeviceModule]; + audio_device_delegate_ = nil; + return -1; + } + } + + playout_parameters_.reset([audio_device_delegate_ preferredOutputSampleRate], 1); + UpdateOutputAudioDeviceBuffer(); + + record_parameters_.reset([audio_device_delegate_ preferredInputSampleRate], 1); + UpdateInputAudioDeviceBuffer(); + + is_initialized_ = true; + + RTC_LOG_F(LS_INFO) << "Did initialize"; + return 0; +} + +int32_t ObjCAudioDeviceModule::Terminate() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (!Initialized()) { + RTC_LOG_F(LS_INFO) << "Not initialized"; + return 0; + } + + if ([audio_device_ isInitialized]) { + if (![audio_device_ terminateDevice]) { + RTC_LOG_F(LS_ERROR) << "Failed to terminate audio device"; + return -1; + } + } + + if (audio_device_delegate_ != nil) { + [audio_device_delegate_ resetAudioDeviceModule]; + audio_device_delegate_ = nil; + } + + is_initialized_ = false; + is_playout_initialized_ = false; + is_recording_initialized_ = false; + thread_ = nullptr; + + RTC_LOG_F(LS_INFO) << "Did terminate"; + return 0; +} + +bool ObjCAudioDeviceModule::Initialized() const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + return is_initialized_ && [audio_device_ isInitialized]; +} + +int32_t ObjCAudioDeviceModule::PlayoutIsAvailable(bool* available) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + *available = Initialized(); + return 0; +} + +bool ObjCAudioDeviceModule::PlayoutIsInitialized() const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + return Initialized() && is_playout_initialized_ && [audio_device_ isPlayoutInitialized]; +} + +int32_t ObjCAudioDeviceModule::InitPlayout() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!Initialized()) { + return -1; + } + if (PlayoutIsInitialized()) { + return 0; + } + RTC_DCHECK(!playing_.load()); + + if (![audio_device_ isPlayoutInitialized]) { + if (![audio_device_ initializePlayout]) { + RTC_LOG_F(LS_ERROR) << "Failed to initialize audio device playout"; + return -1; + } + } + + if (UpdateAudioParameters(playout_parameters_, PlayoutParameters(audio_device_))) { + UpdateOutputAudioDeviceBuffer(); + } + + is_playout_initialized_ = true; + RTC_LOG_F(LS_INFO) << "Did initialize playout"; + return 0; +} + +bool ObjCAudioDeviceModule::Playing() const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + return playing_.load() && [audio_device_ isPlaying]; +} + +int32_t ObjCAudioDeviceModule::StartPlayout() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!PlayoutIsInitialized()) { + return -1; + } + if (Playing()) { + return 0; + } + + audio_device_buffer_->StartPlayout(); + if (playout_fine_audio_buffer_) { + playout_fine_audio_buffer_->ResetPlayout(); + } + if (![audio_device_ startPlayout]) { + RTC_LOG_F(LS_ERROR) << "Failed to start audio device playout"; + return -1; + } + playing_.store(true, std::memory_order_release); + RTC_LOG_F(LS_INFO) << "Did start playout"; + return 0; +} + +int32_t ObjCAudioDeviceModule::StopPlayout() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (![audio_device_ stopPlayout]) { + RTC_LOG_F(LS_WARNING) << "Failed to stop playout"; + return -1; + } + + audio_device_buffer_->StopPlayout(); + playing_.store(false, std::memory_order_release); + RTC_LOG_F(LS_INFO) << "Did stop playout"; + return 0; +} + +int32_t ObjCAudioDeviceModule::PlayoutDelay(uint16_t* delayMS) const { + RTC_DCHECK_RUN_ON(&thread_checker_); + *delayMS = static_cast<uint16_t>(rtc::SafeClamp<int>( + cached_playout_delay_ms_.load(), 0, std::numeric_limits<uint16_t>::max())); + return 0; +} + +int32_t ObjCAudioDeviceModule::RecordingIsAvailable(bool* available) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + *available = Initialized(); + return 0; +} + +bool ObjCAudioDeviceModule::RecordingIsInitialized() const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + return Initialized() && is_recording_initialized_ && [audio_device_ isRecordingInitialized]; +} + +int32_t ObjCAudioDeviceModule::InitRecording() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!Initialized()) { + return -1; + } + if (RecordingIsInitialized()) { + return 0; + } + RTC_DCHECK(!recording_.load()); + + if (![audio_device_ isRecordingInitialized]) { + if (![audio_device_ initializeRecording]) { + RTC_LOG_F(LS_ERROR) << "Failed to initialize audio device recording"; + return -1; + } + } + + if (UpdateAudioParameters(record_parameters_, RecordParameters(audio_device_))) { + UpdateInputAudioDeviceBuffer(); + } + + is_recording_initialized_ = true; + RTC_LOG_F(LS_INFO) << "Did initialize recording"; + return 0; +} + +bool ObjCAudioDeviceModule::Recording() const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + return recording_.load() && [audio_device_ isRecording]; +} + +int32_t ObjCAudioDeviceModule::StartRecording() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!RecordingIsInitialized()) { + return -1; + } + if (Recording()) { + return 0; + } + + audio_device_buffer_->StartRecording(); + if (record_fine_audio_buffer_) { + record_fine_audio_buffer_->ResetRecord(); + } + + if (![audio_device_ startRecording]) { + RTC_LOG_F(LS_ERROR) << "Failed to start audio device recording"; + return -1; + } + recording_.store(true, std::memory_order_release); + RTC_LOG_F(LS_INFO) << "Did start recording"; + return 0; +} + +int32_t ObjCAudioDeviceModule::StopRecording() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (![audio_device_ stopRecording]) { + RTC_LOG_F(LS_WARNING) << "Failed to stop recording"; + return -1; + } + audio_device_buffer_->StopRecording(); + recording_.store(false, std::memory_order_release); + RTC_LOG_F(LS_INFO) << "Did stop recording"; + return 0; +} + +#if defined(WEBRTC_IOS) + +int ObjCAudioDeviceModule::GetPlayoutAudioParameters(AudioParameters* params) const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK(playout_parameters_.is_valid()); + RTC_DCHECK_RUN_ON(&thread_checker_); + *params = playout_parameters_; + return 0; +} + +int ObjCAudioDeviceModule::GetRecordAudioParameters(AudioParameters* params) const { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK(record_parameters_.is_valid()); + RTC_DCHECK_RUN_ON(&thread_checker_); + *params = record_parameters_; + return 0; +} + +#endif // WEBRTC_IOS + +void ObjCAudioDeviceModule::UpdateOutputAudioDeviceBuffer() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first"; + + RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0); + RTC_DCHECK(playout_parameters_.channels() == 1 || playout_parameters_.channels() == 2); + + audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate()); + audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels()); + playout_fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_.get())); +} + +void ObjCAudioDeviceModule::UpdateInputAudioDeviceBuffer() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first"; + + RTC_DCHECK_GT(record_parameters_.sample_rate(), 0); + RTC_DCHECK(record_parameters_.channels() == 1 || record_parameters_.channels() == 2); + + audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate()); + audio_device_buffer_->SetRecordingChannels(record_parameters_.channels()); + record_fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_.get())); +} + +void ObjCAudioDeviceModule::UpdateAudioDelay(std::atomic<int>& delay_ms, + const NSTimeInterval device_latency) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + int latency_ms = static_cast<int>(rtc::kNumMillisecsPerSec * device_latency); + if (latency_ms <= 0) { + return; + } + const int old_latency_ms = delay_ms.exchange(latency_ms); + if (old_latency_ms != latency_ms) { + RTC_LOG_F(LS_INFO) << "Did change audio IO latency from: " << old_latency_ms + << " ms to: " << latency_ms << " ms"; + } +} + +bool ObjCAudioDeviceModule::UpdateAudioParameters(AudioParameters& params, + const AudioParameters& device_params) { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!device_params.is_complete()) { + RTC_LOG_F(LS_INFO) << "Device params are incomplete: " << device_params.ToString(); + return false; + } + if (params.channels() == device_params.channels() && + params.frames_per_buffer() == device_params.frames_per_buffer() && + params.sample_rate() == device_params.sample_rate()) { + RTC_LOG_F(LS_INFO) << "Device params: " << device_params.ToString() + << " are not different from: " << params.ToString(); + return false; + } + + RTC_LOG_F(LS_INFO) << "Audio params will be changed from: " << params.ToString() + << " to: " << device_params.ToString(); + params.reset( + device_params.sample_rate(), device_params.channels(), device_params.frames_per_buffer()); + return true; +} + +OSStatus ObjCAudioDeviceModule::OnDeliverRecordedData( + AudioUnitRenderActionFlags* flags, + const AudioTimeStamp* time_stamp, + NSInteger bus_number, + UInt32 num_frames, + const AudioBufferList* io_data, + void* render_context, + RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) render_block) { + RTC_DCHECK_RUN_ON(&io_record_thread_checker_); + OSStatus result = noErr; + // Simply return if recording is not enabled. + if (!recording_.load()) return result; + + if (io_data != nullptr) { + // AudioBuffer already fullfilled with audio data + RTC_DCHECK_EQ(1, io_data->mNumberBuffers); + const AudioBuffer* audio_buffer = &io_data->mBuffers[0]; + RTC_DCHECK(audio_buffer->mNumberChannels == 1 || audio_buffer->mNumberChannels == 2); + + record_fine_audio_buffer_->DeliverRecordedData( + rtc::ArrayView<const int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames), + cached_recording_delay_ms_.load()); + return noErr; + } + RTC_DCHECK(render_block != nullptr) << "Either io_data or render_block must be provided"; + + // Set the size of our own audio buffer and clear it first to avoid copying + // in combination with potential reallocations. + // On real iOS devices, the size will only be set once (at first callback). + const int channels_count = record_parameters_.channels(); + record_audio_buffer_.Clear(); + record_audio_buffer_.SetSize(num_frames * channels_count); + + // Allocate AudioBuffers to be used as storage for the received audio. + // The AudioBufferList structure works as a placeholder for the + // AudioBuffer structure, which holds a pointer to the actual data buffer + // in `record_audio_buffer_`. Recorded audio will be rendered into this memory + // at each input callback when calling `render_block`. + AudioBufferList audio_buffer_list; + audio_buffer_list.mNumberBuffers = 1; + AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0]; + audio_buffer->mNumberChannels = channels_count; + audio_buffer->mDataByteSize = + record_audio_buffer_.size() * sizeof(decltype(record_audio_buffer_)::value_type); + audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data()); + + // Obtain the recorded audio samples by initiating a rendering cycle into own buffer. + result = + render_block(flags, time_stamp, bus_number, num_frames, &audio_buffer_list, render_context); + if (result != noErr) { + RTC_LOG_F(LS_ERROR) << "Failed to render audio: " << result; + return result; + } + + // Get a pointer to the recorded audio and send it to the WebRTC ADB. + // Use the FineAudioBuffer instance to convert between native buffer size + // and the 10ms buffer size used by WebRTC. + record_fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_, + cached_recording_delay_ms_.load()); + return noErr; +} + +OSStatus ObjCAudioDeviceModule::OnGetPlayoutData(AudioUnitRenderActionFlags* flags, + const AudioTimeStamp* time_stamp, + NSInteger bus_number, + UInt32 num_frames, + AudioBufferList* io_data) { + RTC_DCHECK_RUN_ON(&io_playout_thread_checker_); + // Verify 16-bit, noninterleaved mono or stereo PCM signal format. + RTC_DCHECK_EQ(1, io_data->mNumberBuffers); + AudioBuffer* audio_buffer = &io_data->mBuffers[0]; + RTC_DCHECK(audio_buffer->mNumberChannels == 1 || audio_buffer->mNumberChannels == 2); + RTC_DCHECK_EQ(audio_buffer->mDataByteSize, + sizeof(int16_t) * num_frames * audio_buffer->mNumberChannels); + + // Produce silence and give player a hint about it if playout is not + // activated. + if (!playing_.load()) { + *flags |= kAudioUnitRenderAction_OutputIsSilence; + memset(static_cast<int8_t*>(audio_buffer->mData), 0, audio_buffer->mDataByteSize); + return noErr; + } + + // Read decoded 16-bit PCM samples from WebRTC into the + // `io_data` destination buffer. + playout_fine_audio_buffer_->GetPlayoutData( + rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), + num_frames * audio_buffer->mNumberChannels), + cached_playout_delay_ms_.load()); + + return noErr; +} + +void ObjCAudioDeviceModule::HandleAudioInputInterrupted() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + io_record_thread_checker_.Detach(); +} + +void ObjCAudioDeviceModule::HandleAudioOutputInterrupted() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + io_playout_thread_checker_.Detach(); +} + +void ObjCAudioDeviceModule::HandleAudioInputParametersChange() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (UpdateAudioParameters(record_parameters_, RecordParameters(audio_device_))) { + UpdateInputAudioDeviceBuffer(); + } + + UpdateAudioDelay(cached_recording_delay_ms_, [audio_device_ inputLatency]); +} + +void ObjCAudioDeviceModule::HandleAudioOutputParametersChange() { + RTC_DLOG_F(LS_VERBOSE) << ""; + RTC_DCHECK_RUN_ON(&thread_checker_); + + if (UpdateAudioParameters(playout_parameters_, PlayoutParameters(audio_device_))) { + UpdateOutputAudioDeviceBuffer(); + } + + UpdateAudioDelay(cached_playout_delay_ms_, [audio_device_ outputLatency]); +} + +#pragma mark - Not implemented/Not relevant methods from AudioDeviceModule + +int32_t ObjCAudioDeviceModule::ActiveAudioLayer(AudioLayer* audioLayer) const { + return -1; +} + +int16_t ObjCAudioDeviceModule::PlayoutDevices() { + return 0; +} + +int16_t ObjCAudioDeviceModule::RecordingDevices() { + return 0; +} + +int32_t ObjCAudioDeviceModule::PlayoutDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + return -1; +} + +int32_t ObjCAudioDeviceModule::RecordingDeviceName(uint16_t index, + char name[kAdmMaxDeviceNameSize], + char guid[kAdmMaxGuidSize]) { + return -1; +} + +int32_t ObjCAudioDeviceModule::SetPlayoutDevice(uint16_t index) { + return 0; +} + +int32_t ObjCAudioDeviceModule::SetPlayoutDevice(WindowsDeviceType device) { + return -1; +} + +int32_t ObjCAudioDeviceModule::SetRecordingDevice(uint16_t index) { + return 0; +} + +int32_t ObjCAudioDeviceModule::SetRecordingDevice(WindowsDeviceType device) { + return -1; +} + +int32_t ObjCAudioDeviceModule::InitSpeaker() { + return 0; +} + +bool ObjCAudioDeviceModule::SpeakerIsInitialized() const { + return true; +} + +int32_t ObjCAudioDeviceModule::InitMicrophone() { + return 0; +} + +bool ObjCAudioDeviceModule::MicrophoneIsInitialized() const { + return true; +} + +int32_t ObjCAudioDeviceModule::SpeakerVolumeIsAvailable(bool* available) { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetSpeakerVolume(uint32_t volume) { + return -1; +} + +int32_t ObjCAudioDeviceModule::SpeakerVolume(uint32_t* volume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MaxSpeakerVolume(uint32_t* maxVolume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MinSpeakerVolume(uint32_t* minVolume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::SpeakerMuteIsAvailable(bool* available) { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetSpeakerMute(bool enable) { + return -1; +} + +int32_t ObjCAudioDeviceModule::SpeakerMute(bool* enabled) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MicrophoneMuteIsAvailable(bool* available) { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetMicrophoneMute(bool enable) { + return -1; +} + +int32_t ObjCAudioDeviceModule::MicrophoneMute(bool* enabled) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MicrophoneVolumeIsAvailable(bool* available) { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetMicrophoneVolume(uint32_t volume) { + return -1; +} + +int32_t ObjCAudioDeviceModule::MicrophoneVolume(uint32_t* volume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MaxMicrophoneVolume(uint32_t* maxVolume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::MinMicrophoneVolume(uint32_t* minVolume) const { + return -1; +} + +int32_t ObjCAudioDeviceModule::StereoPlayoutIsAvailable(bool* available) const { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetStereoPlayout(bool enable) { + return -1; +} + +int32_t ObjCAudioDeviceModule::StereoPlayout(bool* enabled) const { + *enabled = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::StereoRecordingIsAvailable(bool* available) const { + *available = false; + return 0; +} + +int32_t ObjCAudioDeviceModule::SetStereoRecording(bool enable) { + return -1; +} + +int32_t ObjCAudioDeviceModule::StereoRecording(bool* enabled) const { + *enabled = false; + return 0; +} + +bool ObjCAudioDeviceModule::BuiltInAECIsAvailable() const { + return false; +} + +int32_t ObjCAudioDeviceModule::EnableBuiltInAEC(bool enable) { + return 0; +} + +bool ObjCAudioDeviceModule::BuiltInAGCIsAvailable() const { + return false; +} + +int32_t ObjCAudioDeviceModule::EnableBuiltInAGC(bool enable) { + return 0; +} + +bool ObjCAudioDeviceModule::BuiltInNSIsAvailable() const { + return false; +} + +int32_t ObjCAudioDeviceModule::EnableBuiltInNS(bool enable) { + return 0; +} + +int32_t ObjCAudioDeviceModule::GetPlayoutUnderrunCount() const { + return -1; +} + +} // namespace objc_adm + +} // namespace webrtc |