blob: 31d5b5ecfc51e9a8ca53b2c3aee89865983e06f4 [file] [log] [blame]
/*
* 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 <stddef.h>
#include "modules/audio_device/audio_device_config.h" // IWYU pragma: keep
#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "system_wrappers/include/metrics.h"
#if defined(_WIN32)
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#include "modules/audio_device/win/audio_device_core_win.h"
#endif
#elif defined(WEBRTC_ANDROID)
#include <stdlib.h>
#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
#include "modules/audio_device/android/aaudio_player.h"
#include "modules/audio_device/android/aaudio_recorder.h"
#endif
#include "modules/audio_device/android/audio_device_template.h"
#include "modules/audio_device/android/audio_manager.h"
#include "modules/audio_device/android/audio_record_jni.h"
#include "modules/audio_device/android/audio_track_jni.h"
#include "modules/audio_device/android/opensles_player.h"
#include "modules/audio_device/android/opensles_recorder.h"
#elif defined(WEBRTC_LINUX)
#if defined(LINUX_ALSA)
#include "modules/audio_device/linux/audio_device_alsa_linux.h"
#endif
#if defined(LINUX_PULSE)
#include "modules/audio_device/linux/audio_device_pulse_linux.h"
#endif
#elif defined(WEBRTC_IOS)
#include "modules/audio_device/ios/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 {
rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
const AudioLayer audio_layer) {
RTC_LOG(INFO) << __FUNCTION__;
return AudioDeviceModule::CreateForTest(audio_layer);
}
// static
rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
const AudioLayer audio_layer) {
RTC_LOG(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;
}
// Create the generic reference counted (platform independent) implementation.
rtc::scoped_refptr<AudioDeviceModuleImpl> audioDevice(
new rtc::RefCountedObject<AudioDeviceModuleImpl>(audio_layer));
// Ensure that the current platform is supported.
if (audioDevice->CheckPlatform() == -1) {
return nullptr;
}
// Create the platform-dependent implementation.
if (audioDevice->CreatePlatformSpecificObjects() == -1) {
return nullptr;
}
// Ensure that the generic audio buffer can communicate with the platform
// specific parts.
if (audioDevice->AttachAudioBuffer() == -1) {
return nullptr;
}
return audioDevice;
}
// TODO(bugs.webrtc.org/7306): deprecated.
rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
const int32_t id,
const AudioLayer audio_layer) {
RTC_LOG(INFO) << __FUNCTION__;
return AudioDeviceModule::Create(audio_layer);
}
AudioDeviceModuleImpl::AudioDeviceModuleImpl(const AudioLayer audioLayer)
: audio_layer_(audioLayer) {
RTC_LOG(INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleImpl::CheckPlatform() {
RTC_LOG(INFO) << __FUNCTION__;
// Ensure that the current platform is supported
PlatformType platform(kPlatformNotSupported);
#if defined(_WIN32)
platform = kPlatformWin32;
RTC_LOG(INFO) << "current platform is Win32";
#elif defined(WEBRTC_ANDROID)
platform = kPlatformAndroid;
RTC_LOG(INFO) << "current platform is Android";
#elif defined(WEBRTC_LINUX)
platform = kPlatformLinux;
RTC_LOG(INFO) << "current platform is Linux";
#elif defined(WEBRTC_IOS)
platform = kPlatformIOS;
RTC_LOG(INFO) << "current platform is IOS";
#elif defined(WEBRTC_MAC)
platform = kPlatformMac;
RTC_LOG(INFO) << "current platform is Mac";
#endif
if (platform == kPlatformNotSupported) {
RTC_LOG(LERROR)
<< "current platform is not supported => this module will self "
"destruct!";
return -1;
}
platform_type_ = platform;
return 0;
}
int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
RTC_LOG(INFO) << __FUNCTION__;
// Dummy ADM implementations if build flags are set.
#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
audio_device_.reset(new AudioDeviceDummy());
RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
if (audio_device_) {
RTC_LOG(INFO) << "Will use file-playing dummy device.";
} else {
// Create a dummy device instead.
audio_device_.reset(new AudioDeviceDummy());
RTC_LOG(INFO) << "Dummy Audio APIs will be utilized";
}
// Real (non-dummy) ADM implementations.
#else
AudioLayer audio_layer(PlatformAudioLayer());
// Windows ADM implementation.
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
if ((audio_layer == kWindowsCoreAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
RTC_LOG(INFO) << "Attempting to use the Windows Core Audio APIs...";
if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
audio_device_.reset(new AudioDeviceWindowsCore());
RTC_LOG(INFO) << "Windows Core Audio APIs will be utilized";
}
}
#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#if defined(WEBRTC_ANDROID)
// Create an Android audio manager.
audio_manager_android_.reset(new AudioManager());
// Select best possible combination of audio layers.
if (audio_layer == kPlatformDefaultAudio) {
if (audio_manager_android_->IsAAudioSupported()) {
// Use of AAudio for both playout and recording has highest priority.
audio_layer = kAndroidAAudioAudio;
} else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
audio_manager_android_->IsLowLatencyRecordSupported()) {
// Use OpenSL ES for both playout and recording.
audio_layer = kAndroidOpenSLESAudio;
} else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
!audio_manager_android_->IsLowLatencyRecordSupported()) {
// Use OpenSL ES for output on devices that only supports the
// low-latency output audio path.
audio_layer = kAndroidJavaInputAndOpenSLESOutputAudio;
} else {
// Use Java-based audio in both directions when low-latency output is
// not supported.
audio_layer = kAndroidJavaAudio;
}
}
AudioManager* audio_manager = audio_manager_android_.get();
if (audio_layer == kAndroidJavaAudio) {
// Java audio for both input and output audio.
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidOpenSLESAudio) {
// OpenSL ES based audio for both input and output audio.
audio_device_.reset(
new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidJavaInputAndOpenSLESOutputAudio) {
// Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
// This combination provides low-latency output audio and at the same
// time support for HW AEC using the AudioRecord Java API.
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
audio_layer, audio_manager));
} else if (audio_layer == kAndroidAAudioAudio) {
#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
// AAudio based audio for both input and output.
audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
audio_layer, audio_manager));
#endif
} else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
// Java audio for input and AAudio for output audio (i.e. mixed APIs).
audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
audio_layer, audio_manager));
#endif
} else {
RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
audio_device_.reset(nullptr);
}
// END #if defined(WEBRTC_ANDROID)
// Linux ADM implementation.
// Note that, LINUX_ALSA is always defined by default when WEBRTC_LINUX is
// defined. 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.
#elif defined(WEBRTC_LINUX)
#if !defined(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(WARNING) << "PulseAudio is disabled using build flag.";
if ((audio_layer == kLinuxAlsaAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
audio_device_.reset(new AudioDeviceLinuxALSA());
RTC_LOG(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(INFO) << "PulseAudio support is enabled.";
if ((audio_layer == kLinuxPulseAudio) ||
(audio_layer == kPlatformDefaultAudio)) {
// Linux PulseAudio implementation is default.
audio_device_.reset(new AudioDeviceLinuxPulse());
RTC_LOG(INFO) << "Linux PulseAudio APIs will be utilized";
} else if (audio_layer == kLinuxAlsaAudio) {
audio_device_.reset(new AudioDeviceLinuxALSA());
RTC_LOG(WARNING) << "Linux ALSA APIs will be utilized.";
}
#endif // #if !defined(LINUX_PULSE)
#endif // #if defined(WEBRTC_LINUX)
// iOS ADM implementation.
#if defined(WEBRTC_IOS)
if (audio_layer == kPlatformDefaultAudio) {
audio_device_.reset(new AudioDeviceIOS());
RTC_LOG(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(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(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(INFO) << __FUNCTION__;
audio_device_->AttachAudioBuffer(&audio_device_buffer_);
return 0;
}
AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
RTC_LOG(INFO) << __FUNCTION__;
}
int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
RTC_LOG(INFO) << __FUNCTION__;
AudioLayer activeAudio;
if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
return -1;
}
*audioLayer = activeAudio;
return 0;
}
int32_t AudioDeviceModuleImpl::Init() {
RTC_LOG(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(INFO) << __FUNCTION__;
if (!initialized_)
return 0;
if (audio_device_->Terminate() == -1) {
return -1;
}
initialized_ = false;
return 0;
}
bool AudioDeviceModuleImpl::Initialized() const {
RTC_LOG(INFO) << __FUNCTION__ << ": " << initialized_;
return initialized_;
}
int32_t AudioDeviceModuleImpl::InitSpeaker() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitSpeaker();
}
int32_t AudioDeviceModuleImpl::InitMicrophone() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->InitMicrophone();
}
int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerVolume(volume);
}
int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->SpeakerVolume(level) == -1) {
return -1;
}
*volume = level;
RTC_LOG(INFO) << "output: " << *volume;
return 0;
}
bool AudioDeviceModuleImpl::SpeakerIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->SpeakerIsInitialized();
RTC_LOG(INFO) << "output: " << isInitialized;
return isInitialized;
}
bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isInitialized = audio_device_->MicrophoneIsInitialized();
RTC_LOG(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(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return audio_device_->SetSpeakerMute(enable);
}
int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->SpeakerMute(muted) == -1) {
return -1;
}
*enabled = muted;
RTC_LOG(INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneMute(enable));
}
int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool muted = false;
if (audio_device_->MicrophoneMute(muted) == -1) {
return -1;
}
*enabled = muted;
RTC_LOG(INFO) << "output: " << muted;
return 0;
}
int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
CHECKinitialized_();
return (audio_device_->SetMicrophoneVolume(volume));
}
int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint32_t level = 0;
if (audio_device_->MicrophoneVolume(level) == -1) {
return -1;
}
*volume = level;
RTC_LOG(INFO) << "output: " << *volume;
return 0;
}
int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable(
bool* available) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->RecordingIsInitialized()) {
RTC_LOG(WARNING) << "recording in stereo is not supported";
return -1;
}
if (audio_device_->SetStereoRecording(enable) == -1) {
RTC_LOG(WARNING) << "failed to change 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(INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoRecording(stereo) == -1) {
return -1;
}
*enabled = stereo;
RTC_LOG(INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
if (audio_device_->PlayoutIsInitialized()) {
RTC_LOG(LERROR)
<< "unable to set stereo mode while playing side is initialized";
return -1;
}
if (audio_device_->SetStereoPlayout(enable)) {
RTC_LOG(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(INFO) << __FUNCTION__;
CHECKinitialized_();
bool stereo = false;
if (audio_device_->StereoPlayout(stereo) == -1) {
return -1;
}
*enabled = stereo;
RTC_LOG(INFO) << "output: " << stereo;
return 0;
}
int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(INFO) << "output: " << isAvailable;
return 0;
}
int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
bool isAvailable = false;
if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
return -1;
}
*available = isAvailable;
RTC_LOG(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(INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
RTC_LOG(INFO) << "output: " << nPlayoutDevices;
return (int16_t)(nPlayoutDevices);
}
int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(index);
}
int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetPlayoutDevice(device);
}
int32_t AudioDeviceModuleImpl::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
}
if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
return -1;
}
if (name != NULL) {
RTC_LOG(INFO) << "output: name = " << name;
}
if (guid != NULL) {
RTC_LOG(INFO) << "output: guid = " << guid;
}
return 0;
}
int32_t AudioDeviceModuleImpl::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
CHECKinitialized_();
if (name == NULL) {
return -1;
}
if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
return -1;
}
if (name != NULL) {
RTC_LOG(INFO) << "output: name = " << name;
}
if (guid != NULL) {
RTC_LOG(INFO) << "output: guid = " << guid;
}
return 0;
}
int16_t AudioDeviceModuleImpl::RecordingDevices() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
uint16_t nRecordingDevices = audio_device_->RecordingDevices();
RTC_LOG(INFO) << "output: " << nRecordingDevices;
return (int16_t)nRecordingDevices;
}
int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
CHECKinitialized_();
return audio_device_->SetRecordingDevice(index);
}
int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
return audio_device_->SetRecordingDevice(device);
}
int32_t AudioDeviceModuleImpl::InitPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (PlayoutIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::InitRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (RecordingIsInitialized()) {
return 0;
}
int32_t result = audio_device_->InitRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::PlayoutIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->PlayoutIsInitialized();
}
bool AudioDeviceModuleImpl::RecordingIsInitialized() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->RecordingIsInitialized();
}
int32_t AudioDeviceModuleImpl::StartPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (Playing()) {
return 0;
}
audio_device_buffer_.StartPlayout();
int32_t result = audio_device_->StartPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::StopPlayout() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopPlayout();
audio_device_buffer_.StopPlayout();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::Playing() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Playing();
}
int32_t AudioDeviceModuleImpl::StartRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
if (Recording()) {
return 0;
}
audio_device_buffer_.StartRecording();
int32_t result = audio_device_->StartRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
int32_t AudioDeviceModuleImpl::StopRecording() {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized_();
int32_t result = audio_device_->StopRecording();
audio_device_buffer_.StopRecording();
RTC_LOG(INFO) << "output: " << result;
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
static_cast<int>(result == 0));
return result;
}
bool AudioDeviceModuleImpl::Recording() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
return audio_device_->Recording();
}
int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
AudioTransport* audioCallback) {
RTC_LOG(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(LERROR) << "failed to retrieve the playout delay";
return -1;
}
*delayMS = delay;
return 0;
}
bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAECIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAEC(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInAGC(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
RTC_LOG(INFO) << __FUNCTION__;
CHECKinitialized__BOOL();
bool isAvailable = audio_device_->BuiltInNSIsAvailable();
RTC_LOG(INFO) << "output: " << isAvailable;
return isAvailable;
}
int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
CHECKinitialized_();
int32_t ok = audio_device_->EnableBuiltInNS(enable);
RTC_LOG(INFO) << "output: " << ok;
return ok;
}
#if defined(WEBRTC_IOS)
int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
AudioParameters* params) const {
RTC_LOG(INFO) << __FUNCTION__;
int r = audio_device_->GetPlayoutAudioParameters(params);
RTC_LOG(INFO) << "output: " << r;
return r;
}
int AudioDeviceModuleImpl::GetRecordAudioParameters(
AudioParameters* params) const {
RTC_LOG(INFO) << __FUNCTION__;
int r = audio_device_->GetRecordAudioParameters(params);
RTC_LOG(INFO) << "output: " << r;
return r;
}
#endif // WEBRTC_IOS
AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
RTC_LOG(INFO) << __FUNCTION__;
return platform_type_;
}
AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
const {
RTC_LOG(INFO) << __FUNCTION__;
return audio_layer_;
}
} // namespace webrtc