| /* |
| * 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 |