| /* | |
| * Copyright (c) 2012 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 "modules/audio_device/audio_device_impl.h" | |
| #include <cstdint> | |
| #include <memory> | |
| #include <utility> | |
| #include "absl/base/nullability.h" | |
| #include "api/audio/audio_device.h" | |
| #include "api/audio/audio_device_defines.h" | |
| #include "api/environment/environment.h" | |
| #include "api/make_ref_counted.h" | |
| #include "api/scoped_refptr.h" | |
| #include "modules/audio_device/audio_device_generic.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/logging.h" | |
| #include "system_wrappers/include/metrics.h" | |
| #if defined(WEBRTC_WIN) | |
| #include "modules/audio_device/win/audio_device_core_win.h" | |
| #elif defined(WEBRTC_LINUX) | |
| #if defined(WEBRTC_ENABLE_LINUX_ALSA) | |
| #include "modules/audio_device/linux/audio_device_alsa_linux.h" | |
| #endif | |
| #if defined(WEBRTC_ENABLE_LINUX_PULSE) | |
| #include "modules/audio_device/linux/audio_device_pulse_linux.h" | |
| #endif | |
| #elif defined(WEBRTC_IOS) | |
| #include "sdk/objc/native/src/audio/audio_device_ios.h" | |
| #elif defined(WEBRTC_MAC) | |
| #include "modules/audio_device/mac/audio_device_mac.h" | |
| #endif | |
| #if defined(WEBRTC_DUMMY_FILE_DEVICES) | |
| #include "modules/audio_device/dummy/file_audio_device.h" | |
| #include "modules/audio_device/dummy/file_audio_device_factory.h" | |
| #endif | |
| #include "modules/audio_device/dummy/audio_device_dummy.h" | |
| #define CHECKinitialized_() \ | |
| { \ | |
| if (!initialized_) { \ | |
| return -1; \ | |
| } \ | |
| } | |
| #define CHECKinitialized__BOOL() \ | |
| { \ | |
| if (!initialized_) { \ | |
| return false; \ | |
| } \ | |
| } | |
| namespace webrtc { | |
| absl_nullable scoped_refptr<AudioDeviceModuleImpl> | |
| AudioDeviceModuleImpl::Create(const Environment& env, AudioLayer audio_layer) { | |
| RTC_DLOG(LS_INFO) << __FUNCTION__; | |
| // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own | |
| // dedicated factory method which should be used instead. | |
| if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) { | |
| RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() " | |
| "factory method instead for this option."; | |
| return nullptr; | |
| } else if (audio_layer == AudioDeviceModule::kAndroidJavaAudio || | |
| audio_layer == AudioDeviceModule::kAndroidOpenSLESAudio || | |
| audio_layer == | |
| AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio || | |
| audio_layer == kAndroidAAudioAudio || | |
| audio_layer == kAndroidJavaInputAndAAudioOutputAudio) { | |
| RTC_LOG(LS_ERROR) << "Use the CreateAndroidAudioDeviceModule() " | |
| "factory method instead for this option."; | |
| return nullptr; | |
| } | |
| // Create the generic reference counted (platform independent) implementation. | |
| auto audio_device = make_ref_counted<AudioDeviceModuleImpl>(env, audio_layer); | |
| // Ensure that the current platform is supported. | |
| if (audio_device->CheckPlatform() == -1) { | |
| return nullptr; | |
| } | |
| // Create the platform-dependent implementation. | |
| if (audio_device->CreatePlatformSpecificObjects(env) == -1) { | |
| return nullptr; | |
| } | |
| // Ensure that the generic audio buffer can communicate with the platform | |
| // specific parts. | |
| if (audio_device->AttachAudioBuffer() == -1) { | |
| return nullptr; | |
| } | |
| return audio_device; | |
| } | |
| AudioDeviceModuleImpl::AudioDeviceModuleImpl(const Environment& env, | |
| AudioLayer audio_layer) | |
| : audio_layer_(audio_layer), audio_device_buffer_(env) { | |
| RTC_DLOG(LS_INFO) << __FUNCTION__; | |
| } | |
| AudioDeviceModuleImpl::AudioDeviceModuleImpl( | |
| const Environment& env, | |
| AudioLayer audio_layer, | |
| std::unique_ptr<AudioDeviceGeneric> audio_device, | |
| bool create_detached) | |
| : audio_layer_(audio_layer), | |
| audio_device_buffer_(env, create_detached), | |
| audio_device_(std::move(audio_device)) { | |
| RTC_DLOG(LS_INFO) << __FUNCTION__; | |
| } | |
| int32_t AudioDeviceModuleImpl::CheckPlatform() { | |
| RTC_DLOG(LS_INFO) << __FUNCTION__; | |
| // Ensure that the current platform is supported | |
| PlatformType platform(kPlatformNotSupported); | |
| #if defined(_WIN32) | |
| platform = kPlatformWin32; | |
| RTC_LOG(LS_INFO) << "current platform is Win32"; | |
| #elif defined(WEBRTC_ANDROID) | |
| platform = kPlatformAndroid; | |
| RTC_LOG(LS_INFO) << "current platform is Android"; | |
| #elif defined(WEBRTC_LINUX) | |
| platform = kPlatformLinux; | |
| RTC_LOG(LS_INFO) << "current platform is Linux"; | |
| #elif defined(WEBRTC_IOS) | |
| platform = kPlatformIOS; | |
| RTC_LOG(LS_INFO) << "current platform is IOS"; | |
| #elif defined(WEBRTC_MAC) | |
| platform = kPlatformMac; | |
| RTC_LOG(LS_INFO) << "current platform is Mac"; | |
| #elif defined(WEBRTC_FUCHSIA) | |
| platform = kPlatformFuchsia; | |
| RTC_LOG(LS_INFO) << "current platform is Fuchsia"; | |
| #endif | |
| if (platform == kPlatformNotSupported) { | |
| RTC_LOG(LS_ERROR) | |
| << "current platform is not supported => this module will self " | |
| "destruct!"; | |
| return -1; | |
| } | |
| platform_type_ = platform; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects( | |
| [[maybe_unused]] const Environment& env) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| if (audio_device_ != nullptr) { | |
| RTC_LOG(LS_INFO) << "Reusing provided audio device"; | |
| return 0; | |
| } | |
| // Dummy ADM implementations if build flags are set. | |
| #if defined(WEBRTC_DUMMY_AUDIO_BUILD) | |
| audio_device_.reset(new AudioDeviceDummy()); | |
| RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized"; | |
| #elif defined(WEBRTC_DUMMY_FILE_DEVICES) | |
| audio_device_ = FileAudioDeviceFactory::CreateFileAudioDevice(env); | |
| if (audio_device_) { | |
| RTC_LOG(LS_INFO) << "Will use file-playing dummy device."; | |
| } else { | |
| // Create a dummy device instead. | |
| audio_device_.reset(new AudioDeviceDummy()); | |
| RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized"; | |
| } | |
| // Real (non-dummy) ADM implementations. | |
| #else | |
| AudioLayer audio_layer(PlatformAudioLayer()); | |
| #if defined(WEBRTC_WIN) | |
| // Windows ADM implementation. | |
| if ((audio_layer == kWindowsCoreAudio) || | |
| (audio_layer == kPlatformDefaultAudio)) { | |
| RTC_LOG(LS_INFO) << "Attempting to use the Windows Core Audio APIs..."; | |
| if (AudioDeviceWindowsCore::CoreAudioIsSupported()) { | |
| audio_device_.reset(new AudioDeviceWindowsCore()); | |
| RTC_LOG(LS_INFO) << "Windows Core Audio APIs will be utilized"; | |
| } | |
| } | |
| #endif | |
| // Linux ADM implementation. | |
| // Note that, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when | |
| // WEBRTC_LINUX is defined. WEBRTC_ENABLE_LINUX_PULSE depends on the | |
| // 'rtc_include_pulse_audio' build flag. | |
| // TODO(bugs.webrtc.org/9127): improve support and make it more clear that | |
| // PulseAudio is the default selection. | |
| #if !defined(WEBRTC_ANDROID) && defined(WEBRTC_LINUX) | |
| #if !defined(WEBRTC_ENABLE_LINUX_PULSE) | |
| // Build flag 'rtc_include_pulse_audio' is set to false. In this mode: | |
| // - kPlatformDefaultAudio => ALSA, and | |
| // - kLinuxAlsaAudio => ALSA, and | |
| // - kLinuxPulseAudio => Invalid selection. | |
| RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag."; | |
| if ((audio_layer == kLinuxAlsaAudio) || | |
| (audio_layer == kPlatformDefaultAudio)) { | |
| audio_device_.reset(new AudioDeviceLinuxALSA()); | |
| RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized."; | |
| } | |
| #else | |
| // Build flag 'rtc_include_pulse_audio' is set to true (default). In this | |
| // mode: | |
| // - kPlatformDefaultAudio => PulseAudio, and | |
| // - kLinuxPulseAudio => PulseAudio, and | |
| // - kLinuxAlsaAudio => ALSA (supported but not default). | |
| RTC_LOG(LS_INFO) << "PulseAudio support is enabled."; | |
| if ((audio_layer == kLinuxPulseAudio) || | |
| (audio_layer == kPlatformDefaultAudio)) { | |
| // Linux PulseAudio implementation is default. | |
| audio_device_.reset(new AudioDeviceLinuxPulse()); | |
| RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized"; | |
| } else if (audio_layer == kLinuxAlsaAudio) { | |
| audio_device_.reset(new AudioDeviceLinuxALSA()); | |
| RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized."; | |
| } | |
| #endif // #if !defined(WEBRTC_ENABLE_LINUX_PULSE) | |
| #endif // #if defined(WEBRTC_LINUX) | |
| // iOS ADM implementation. | |
| #if defined(WEBRTC_IOS) | |
| if (audio_layer == kPlatformDefaultAudio) { | |
| audio_device_ = std::make_unique<ios_adm::AudioDeviceIOS>( | |
| env, | |
| /*bypass_voice_processing=*/false, | |
| /*muted_speech_event_handler=*/nullptr, | |
| /*render_error_handler=*/nullptr); | |
| RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized."; | |
| } | |
| // END #if defined(WEBRTC_IOS) | |
| // Mac OS X ADM implementation. | |
| #elif defined(WEBRTC_MAC) | |
| if (audio_layer == kPlatformDefaultAudio) { | |
| audio_device_.reset(new AudioDeviceMac()); | |
| RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized."; | |
| } | |
| #endif // WEBRTC_MAC | |
| // Dummy ADM implementation. | |
| if (audio_layer == kDummyAudio) { | |
| audio_device_.reset(new AudioDeviceDummy()); | |
| RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized."; | |
| } | |
| #endif // if defined(WEBRTC_DUMMY_AUDIO_BUILD) | |
| if (!audio_device_) { | |
| RTC_LOG(LS_ERROR) | |
| << "Failed to create the platform specific ADM implementation."; | |
| return -1; | |
| } | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::AttachAudioBuffer() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| audio_device_->AttachAudioBuffer(&audio_device_buffer_); | |
| return 0; | |
| } | |
| AudioDeviceModuleImpl::~AudioDeviceModuleImpl() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| } | |
| int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| AudioLayer activeAudio; | |
| if (audio_device_->ActiveAudioLayer(activeAudio) == -1) { | |
| return -1; | |
| } | |
| *audioLayer = activeAudio; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::Init() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| if (initialized_) | |
| return 0; | |
| RTC_CHECK(audio_device_); | |
| AudioDeviceGeneric::InitStatus status = audio_device_->Init(); | |
| RTC_HISTOGRAM_ENUMERATION( | |
| "WebRTC.Audio.InitializationResult", static_cast<int>(status), | |
| static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES)); | |
| if (status != AudioDeviceGeneric::InitStatus::OK) { | |
| RTC_LOG(LS_ERROR) << "Audio device initialization failed."; | |
| return -1; | |
| } | |
| initialized_ = true; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::Terminate() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| if (!initialized_) | |
| return 0; | |
| if (audio_device_->Terminate() == -1) { | |
| return -1; | |
| } | |
| initialized_ = false; | |
| return 0; | |
| } | |
| bool AudioDeviceModuleImpl::Initialized() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << ": " << initialized_; | |
| return initialized_; | |
| } | |
| int32_t AudioDeviceModuleImpl::InitSpeaker() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| return audio_device_->InitSpeaker(); | |
| } | |
| int32_t AudioDeviceModuleImpl::InitMicrophone() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| return audio_device_->InitMicrophone(); | |
| } | |
| int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")"; | |
| CHECKinitialized_(); | |
| return audio_device_->SetSpeakerVolume(volume); | |
| } | |
| int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| uint32_t level = 0; | |
| if (audio_device_->SpeakerVolume(level) == -1) { | |
| return -1; | |
| } | |
| *volume = level; | |
| RTC_LOG(LS_INFO) << "output: " << *volume; | |
| return 0; | |
| } | |
| bool AudioDeviceModuleImpl::SpeakerIsInitialized() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| bool isInitialized = audio_device_->SpeakerIsInitialized(); | |
| RTC_LOG(LS_INFO) << "output: " << isInitialized; | |
| return isInitialized; | |
| } | |
| bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| bool isInitialized = audio_device_->MicrophoneIsInitialized(); | |
| RTC_LOG(LS_INFO) << "output: " << isInitialized; | |
| return isInitialized; | |
| } | |
| int32_t AudioDeviceModuleImpl::MaxSpeakerVolume(uint32_t* maxVolume) const { | |
| CHECKinitialized_(); | |
| uint32_t maxVol = 0; | |
| if (audio_device_->MaxSpeakerVolume(maxVol) == -1) { | |
| return -1; | |
| } | |
| *maxVolume = maxVol; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::MinSpeakerVolume(uint32_t* minVolume) const { | |
| CHECKinitialized_(); | |
| uint32_t minVol = 0; | |
| if (audio_device_->MinSpeakerVolume(minVol) == -1) { | |
| return -1; | |
| } | |
| *minVolume = minVol; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| return audio_device_->SetSpeakerMute(enable); | |
| } | |
| int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool muted = false; | |
| if (audio_device_->SpeakerMute(muted) == -1) { | |
| return -1; | |
| } | |
| *enabled = muted; | |
| RTC_LOG(LS_INFO) << "output: " << muted; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| return (audio_device_->SetMicrophoneMute(enable)); | |
| } | |
| int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool muted = false; | |
| if (audio_device_->MicrophoneMute(muted) == -1) { | |
| return -1; | |
| } | |
| *enabled = muted; | |
| RTC_LOG(LS_INFO) << "output: " << muted; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")"; | |
| CHECKinitialized_(); | |
| return (audio_device_->SetMicrophoneVolume(volume)); | |
| } | |
| int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| uint32_t level = 0; | |
| if (audio_device_->MicrophoneVolume(level) == -1) { | |
| return -1; | |
| } | |
| *volume = level; | |
| RTC_LOG(LS_INFO) << "output: " << *volume; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable( | |
| bool* available) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| if (audio_device_->RecordingIsInitialized()) { | |
| RTC_LOG(LS_ERROR) | |
| << "unable to set stereo mode after recording is initialized"; | |
| return -1; | |
| } | |
| if (audio_device_->SetStereoRecording(enable) == -1) { | |
| if (enable) { | |
| RTC_LOG(LS_WARNING) << "failed to enable stereo recording"; | |
| } | |
| return -1; | |
| } | |
| int8_t nChannels(1); | |
| if (enable) { | |
| nChannels = 2; | |
| } | |
| audio_device_buffer_.SetRecordingChannels(nChannels); | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool stereo = false; | |
| if (audio_device_->StereoRecording(stereo) == -1) { | |
| return -1; | |
| } | |
| *enabled = stereo; | |
| RTC_LOG(LS_INFO) << "output: " << stereo; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| if (audio_device_->PlayoutIsInitialized()) { | |
| RTC_LOG(LS_ERROR) | |
| << "unable to set stereo mode while playing side is initialized"; | |
| return -1; | |
| } | |
| if (audio_device_->SetStereoPlayout(enable)) { | |
| RTC_LOG(LS_WARNING) << "stereo playout is not supported"; | |
| return -1; | |
| } | |
| int8_t nChannels(1); | |
| if (enable) { | |
| nChannels = 2; | |
| } | |
| audio_device_buffer_.SetPlayoutChannels(nChannels); | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool stereo = false; | |
| if (audio_device_->StereoPlayout(stereo) == -1) { | |
| return -1; | |
| } | |
| *enabled = stereo; | |
| RTC_LOG(LS_INFO) << "output: " << stereo; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| bool isAvailable = false; | |
| if (audio_device_->RecordingIsAvailable(isAvailable) == -1) { | |
| return -1; | |
| } | |
| *available = isAvailable; | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::MaxMicrophoneVolume(uint32_t* maxVolume) const { | |
| CHECKinitialized_(); | |
| uint32_t maxVol(0); | |
| if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) { | |
| return -1; | |
| } | |
| *maxVolume = maxVol; | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::MinMicrophoneVolume(uint32_t* minVolume) const { | |
| CHECKinitialized_(); | |
| uint32_t minVol(0); | |
| if (audio_device_->MinMicrophoneVolume(minVol) == -1) { | |
| return -1; | |
| } | |
| *minVolume = minVol; | |
| return 0; | |
| } | |
| int16_t AudioDeviceModuleImpl::PlayoutDevices() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| uint16_t nPlayoutDevices = audio_device_->PlayoutDevices(); | |
| RTC_LOG(LS_INFO) << "output: " << nPlayoutDevices; | |
| return (int16_t)(nPlayoutDevices); | |
| } | |
| int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")"; | |
| CHECKinitialized_(); | |
| return audio_device_->SetPlayoutDevice(index); | |
| } | |
| int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| return audio_device_->SetPlayoutDevice(device); | |
| } | |
| int32_t AudioDeviceModuleImpl::PlayoutDeviceName( | |
| uint16_t index, | |
| char name[kAdmMaxDeviceNameSize], | |
| char guid[kAdmMaxGuidSize]) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)"; | |
| CHECKinitialized_(); | |
| if (name == nullptr) { | |
| return -1; | |
| } | |
| if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) { | |
| return -1; | |
| } | |
| if (name != nullptr) { | |
| RTC_LOG(LS_INFO) << "output: name = " << name; | |
| } | |
| if (guid != nullptr) { | |
| RTC_LOG(LS_INFO) << "output: guid = " << guid; | |
| } | |
| return 0; | |
| } | |
| int32_t AudioDeviceModuleImpl::RecordingDeviceName( | |
| uint16_t index, | |
| char name[kAdmMaxDeviceNameSize], | |
| char guid[kAdmMaxGuidSize]) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)"; | |
| CHECKinitialized_(); | |
| if (name == nullptr) { | |
| return -1; | |
| } | |
| if (audio_device_->RecordingDeviceName(index, name, guid) == -1) { | |
| return -1; | |
| } | |
| if (name != nullptr) { | |
| RTC_LOG(LS_INFO) << "output: name = " << name; | |
| } | |
| if (guid != nullptr) { | |
| RTC_LOG(LS_INFO) << "output: guid = " << guid; | |
| } | |
| return 0; | |
| } | |
| int16_t AudioDeviceModuleImpl::RecordingDevices() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| uint16_t nRecordingDevices = audio_device_->RecordingDevices(); | |
| RTC_LOG(LS_INFO) << "output: " << nRecordingDevices; | |
| return (int16_t)nRecordingDevices; | |
| } | |
| int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")"; | |
| CHECKinitialized_(); | |
| return audio_device_->SetRecordingDevice(index); | |
| } | |
| int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| return audio_device_->SetRecordingDevice(device); | |
| } | |
| int32_t AudioDeviceModuleImpl::InitPlayout() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| if (PlayoutIsInitialized()) { | |
| return 0; | |
| } | |
| int32_t result = audio_device_->InitPlayout(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| int32_t AudioDeviceModuleImpl::InitRecording() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| if (RecordingIsInitialized()) { | |
| return 0; | |
| } | |
| int32_t result = audio_device_->InitRecording(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| bool AudioDeviceModuleImpl::PlayoutIsInitialized() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| return audio_device_->PlayoutIsInitialized(); | |
| } | |
| bool AudioDeviceModuleImpl::RecordingIsInitialized() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| return audio_device_->RecordingIsInitialized(); | |
| } | |
| int32_t AudioDeviceModuleImpl::StartPlayout() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| if (Playing()) { | |
| return 0; | |
| } | |
| audio_device_buffer_.StartPlayout(); | |
| int32_t result = audio_device_->StartPlayout(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| int32_t AudioDeviceModuleImpl::StopPlayout() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| int32_t result = audio_device_->StopPlayout(); | |
| audio_device_buffer_.StopPlayout(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| bool AudioDeviceModuleImpl::Playing() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| return audio_device_->Playing(); | |
| } | |
| int32_t AudioDeviceModuleImpl::StartRecording() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| if (Recording()) { | |
| return 0; | |
| } | |
| audio_device_buffer_.StartRecording(); | |
| int32_t result = audio_device_->StartRecording(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| int32_t AudioDeviceModuleImpl::StopRecording() { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized_(); | |
| int32_t result = audio_device_->StopRecording(); | |
| audio_device_buffer_.StopRecording(); | |
| RTC_LOG(LS_INFO) << "output: " << result; | |
| RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess", | |
| static_cast<int>(result == 0)); | |
| return result; | |
| } | |
| bool AudioDeviceModuleImpl::Recording() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| return audio_device_->Recording(); | |
| } | |
| int32_t AudioDeviceModuleImpl::RegisterAudioCallback( | |
| AudioTransport* audioCallback) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| return audio_device_buffer_.RegisterAudioCallback(audioCallback); | |
| } | |
| int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const { | |
| CHECKinitialized_(); | |
| uint16_t delay = 0; | |
| if (audio_device_->PlayoutDelay(delay) == -1) { | |
| RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay"; | |
| return -1; | |
| } | |
| *delayMS = delay; | |
| return 0; | |
| } | |
| bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| bool isAvailable = audio_device_->BuiltInAECIsAvailable(); | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return isAvailable; | |
| } | |
| int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| int32_t ok = audio_device_->EnableBuiltInAEC(enable); | |
| RTC_LOG(LS_INFO) << "output: " << ok; | |
| return ok; | |
| } | |
| bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| bool isAvailable = audio_device_->BuiltInAGCIsAvailable(); | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return isAvailable; | |
| } | |
| int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| int32_t ok = audio_device_->EnableBuiltInAGC(enable); | |
| RTC_LOG(LS_INFO) << "output: " << ok; | |
| return ok; | |
| } | |
| bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| CHECKinitialized__BOOL(); | |
| bool isAvailable = audio_device_->BuiltInNSIsAvailable(); | |
| RTC_LOG(LS_INFO) << "output: " << isAvailable; | |
| return isAvailable; | |
| } | |
| int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) { | |
| RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")"; | |
| CHECKinitialized_(); | |
| int32_t ok = audio_device_->EnableBuiltInNS(enable); | |
| RTC_LOG(LS_INFO) << "output: " << ok; | |
| return ok; | |
| } | |
| int32_t AudioDeviceModuleImpl::GetPlayoutUnderrunCount() const { | |
| CHECKinitialized_(); | |
| int32_t underrunCount = audio_device_->GetPlayoutUnderrunCount(); | |
| return underrunCount; | |
| } | |
| #if defined(WEBRTC_IOS) | |
| int AudioDeviceModuleImpl::GetPlayoutAudioParameters( | |
| AudioParameters* params) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| int r = audio_device_->GetPlayoutAudioParameters(params); | |
| RTC_LOG(LS_INFO) << "output: " << r; | |
| return r; | |
| } | |
| int AudioDeviceModuleImpl::GetRecordAudioParameters( | |
| AudioParameters* params) const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| int r = audio_device_->GetRecordAudioParameters(params); | |
| RTC_LOG(LS_INFO) << "output: " << r; | |
| return r; | |
| } | |
| #endif // WEBRTC_IOS | |
| AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| return platform_type_; | |
| } | |
| AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer() | |
| const { | |
| RTC_LOG(LS_INFO) << __FUNCTION__; | |
| return audio_layer_; | |
| } | |
| } // namespace webrtc |