| /* |
| * Copyright (c) 2015 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/android/audio_manager.h" |
| |
| #include <utility> |
| |
| #include "modules/audio_device/android/audio_common.h" |
| #include "modules/utility/include/helpers_android.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/platform_thread.h" |
| |
| namespace webrtc { |
| |
| // AudioManager::JavaAudioManager implementation |
| AudioManager::JavaAudioManager::JavaAudioManager( |
| NativeRegistration* native_reg, |
| std::unique_ptr<GlobalRef> audio_manager) |
| : audio_manager_(std::move(audio_manager)), |
| init_(native_reg->GetMethodId("init", "()Z")), |
| dispose_(native_reg->GetMethodId("dispose", "()V")), |
| is_communication_mode_enabled_( |
| native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")), |
| is_device_blacklisted_for_open_sles_usage_( |
| native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage", |
| "()Z")) { |
| RTC_LOG(INFO) << "JavaAudioManager::ctor"; |
| } |
| |
| AudioManager::JavaAudioManager::~JavaAudioManager() { |
| RTC_LOG(INFO) << "JavaAudioManager::~dtor"; |
| } |
| |
| bool AudioManager::JavaAudioManager::Init() { |
| return audio_manager_->CallBooleanMethod(init_); |
| } |
| |
| void AudioManager::JavaAudioManager::Close() { |
| audio_manager_->CallVoidMethod(dispose_); |
| } |
| |
| bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() { |
| return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_); |
| } |
| |
| bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() { |
| return audio_manager_->CallBooleanMethod( |
| is_device_blacklisted_for_open_sles_usage_); |
| } |
| |
| // AudioManager implementation |
| AudioManager::AudioManager() |
| : j_environment_(JVM::GetInstance()->environment()), |
| audio_layer_(AudioDeviceModule::kPlatformDefaultAudio), |
| initialized_(false), |
| hardware_aec_(false), |
| hardware_agc_(false), |
| hardware_ns_(false), |
| low_latency_playout_(false), |
| low_latency_record_(false), |
| delay_estimate_in_milliseconds_(0) { |
| RTC_LOG(INFO) << "ctor"; |
| RTC_CHECK(j_environment_); |
| JNINativeMethod native_methods[] = { |
| {"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V", |
| reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}}; |
| j_native_registration_ = j_environment_->RegisterNatives( |
| "org/webrtc/voiceengine/WebRtcAudioManager", native_methods, |
| arraysize(native_methods)); |
| j_audio_manager_.reset( |
| new JavaAudioManager(j_native_registration_.get(), |
| j_native_registration_->NewObject( |
| "<init>", "(J)V", PointerTojlong(this)))); |
| } |
| |
| AudioManager::~AudioManager() { |
| RTC_LOG(INFO) << "dtor"; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| Close(); |
| } |
| |
| void AudioManager::SetActiveAudioLayer( |
| AudioDeviceModule::AudioLayer audio_layer) { |
| RTC_LOG(INFO) << "SetActiveAudioLayer: " << audio_layer; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| RTC_DCHECK(!initialized_); |
| // Store the currently utilized audio layer. |
| audio_layer_ = audio_layer; |
| // The delay estimate can take one of two fixed values depending on if the |
| // device supports low-latency output or not. However, it is also possible |
| // that the user explicitly selects the high-latency audio path, hence we use |
| // the selected |audio_layer| here to set the delay estimate. |
| delay_estimate_in_milliseconds_ = |
| (audio_layer == AudioDeviceModule::kAndroidJavaAudio) |
| ? kHighLatencyModeDelayEstimateInMilliseconds |
| : kLowLatencyModeDelayEstimateInMilliseconds; |
| RTC_LOG(INFO) << "delay_estimate_in_milliseconds: " |
| << delay_estimate_in_milliseconds_; |
| } |
| |
| SLObjectItf AudioManager::GetOpenSLEngine() { |
| RTC_LOG(INFO) << "GetOpenSLEngine"; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| // Only allow usage of OpenSL ES if such an audio layer has been specified. |
| if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio && |
| audio_layer_ != |
| AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) { |
| RTC_LOG(INFO) |
| << "Unable to create OpenSL engine for the current audio layer: " |
| << audio_layer_; |
| return nullptr; |
| } |
| // OpenSL ES for Android only supports a single engine per application. |
| // If one already has been created, return existing object instead of |
| // creating a new. |
| if (engine_object_.Get() != nullptr) { |
| RTC_LOG(WARNING) << "The OpenSL ES engine object has already been created"; |
| return engine_object_.Get(); |
| } |
| // Create the engine object in thread safe mode. |
| const SLEngineOption option[] = { |
| {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}}; |
| SLresult result = |
| slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL); |
| if (result != SL_RESULT_SUCCESS) { |
| RTC_LOG(LS_ERROR) << "slCreateEngine() failed: " |
| << GetSLErrorString(result); |
| engine_object_.Reset(); |
| return nullptr; |
| } |
| // Realize the SL Engine in synchronous mode. |
| result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE); |
| if (result != SL_RESULT_SUCCESS) { |
| RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result); |
| engine_object_.Reset(); |
| return nullptr; |
| } |
| // Finally return the SLObjectItf interface of the engine object. |
| return engine_object_.Get(); |
| } |
| |
| bool AudioManager::Init() { |
| RTC_LOG(INFO) << "Init"; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| RTC_DCHECK(!initialized_); |
| RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio); |
| if (!j_audio_manager_->Init()) { |
| RTC_LOG(LS_ERROR) << "Init() failed"; |
| return false; |
| } |
| initialized_ = true; |
| return true; |
| } |
| |
| bool AudioManager::Close() { |
| RTC_LOG(INFO) << "Close"; |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| if (!initialized_) |
| return true; |
| j_audio_manager_->Close(); |
| initialized_ = false; |
| return true; |
| } |
| |
| bool AudioManager::IsCommunicationModeEnabled() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return j_audio_manager_->IsCommunicationModeEnabled(); |
| } |
| |
| bool AudioManager::IsAcousticEchoCancelerSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return hardware_aec_; |
| } |
| |
| bool AudioManager::IsAutomaticGainControlSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return hardware_agc_; |
| } |
| |
| bool AudioManager::IsNoiseSuppressorSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return hardware_ns_; |
| } |
| |
| bool AudioManager::IsLowLatencyPlayoutSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| // Some devices are blacklisted for usage of OpenSL ES even if they report |
| // that low-latency playout is supported. See b/21485703 for details. |
| return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage() |
| ? false |
| : low_latency_playout_; |
| } |
| |
| bool AudioManager::IsLowLatencyRecordSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return low_latency_record_; |
| } |
| |
| bool AudioManager::IsProAudioSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| // TODO(henrika): return the state independently of if OpenSL ES is |
| // blacklisted or not for now. We could use the same approach as in |
| // IsLowLatencyPlayoutSupported() but I can't see the need for it yet. |
| return pro_audio_; |
| } |
| |
| // TODO(henrika): improve comments... |
| bool AudioManager::IsAAudioSupported() const { |
| #if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) |
| return a_audio_; |
| #else |
| return false; |
| #endif |
| } |
| |
| bool AudioManager::IsStereoPlayoutSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (playout_parameters_.channels() == 2); |
| } |
| |
| bool AudioManager::IsStereoRecordSupported() const { |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return (record_parameters_.channels() == 2); |
| } |
| |
| int AudioManager::GetDelayEstimateInMilliseconds() const { |
| return delay_estimate_in_milliseconds_; |
| } |
| |
| JNI_FUNCTION_ALIGN |
| void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env, |
| jobject obj, |
| jint sample_rate, |
| jint output_channels, |
| jint input_channels, |
| jboolean hardware_aec, |
| jboolean hardware_agc, |
| jboolean hardware_ns, |
| jboolean low_latency_output, |
| jboolean low_latency_input, |
| jboolean pro_audio, |
| jboolean a_audio, |
| jint output_buffer_size, |
| jint input_buffer_size, |
| jlong native_audio_manager) { |
| webrtc::AudioManager* this_object = |
| reinterpret_cast<webrtc::AudioManager*>(native_audio_manager); |
| this_object->OnCacheAudioParameters( |
| env, sample_rate, output_channels, input_channels, hardware_aec, |
| hardware_agc, hardware_ns, low_latency_output, low_latency_input, |
| pro_audio, a_audio, output_buffer_size, input_buffer_size); |
| } |
| |
| void AudioManager::OnCacheAudioParameters(JNIEnv* env, |
| jint sample_rate, |
| jint output_channels, |
| jint input_channels, |
| jboolean hardware_aec, |
| jboolean hardware_agc, |
| jboolean hardware_ns, |
| jboolean low_latency_output, |
| jboolean low_latency_input, |
| jboolean pro_audio, |
| jboolean a_audio, |
| jint output_buffer_size, |
| jint input_buffer_size) { |
| RTC_LOG(INFO) |
| << "OnCacheAudioParameters: " |
| "hardware_aec: " |
| << static_cast<bool>(hardware_aec) |
| << ", hardware_agc: " << static_cast<bool>(hardware_agc) |
| << ", hardware_ns: " << static_cast<bool>(hardware_ns) |
| << ", low_latency_output: " << static_cast<bool>(low_latency_output) |
| << ", low_latency_input: " << static_cast<bool>(low_latency_input) |
| << ", pro_audio: " << static_cast<bool>(pro_audio) |
| << ", a_audio: " << static_cast<bool>(a_audio) |
| << ", sample_rate: " << static_cast<int>(sample_rate) |
| << ", output_channels: " << static_cast<int>(output_channels) |
| << ", input_channels: " << static_cast<int>(input_channels) |
| << ", output_buffer_size: " << static_cast<int>(output_buffer_size) |
| << ", input_buffer_size: " << static_cast<int>(input_buffer_size); |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| hardware_aec_ = hardware_aec; |
| hardware_agc_ = hardware_agc; |
| hardware_ns_ = hardware_ns; |
| low_latency_playout_ = low_latency_output; |
| low_latency_record_ = low_latency_input; |
| pro_audio_ = pro_audio; |
| a_audio_ = a_audio; |
| playout_parameters_.reset(sample_rate, static_cast<size_t>(output_channels), |
| static_cast<size_t>(output_buffer_size)); |
| record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels), |
| static_cast<size_t>(input_buffer_size)); |
| } |
| |
| const AudioParameters& AudioManager::GetPlayoutAudioParameters() { |
| RTC_CHECK(playout_parameters_.is_valid()); |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return playout_parameters_; |
| } |
| |
| const AudioParameters& AudioManager::GetRecordAudioParameters() { |
| RTC_CHECK(record_parameters_.is_valid()); |
| RTC_DCHECK(thread_checker_.IsCurrent()); |
| return record_parameters_; |
| } |
| |
| } // namespace webrtc |
