| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_ |
| |
| #include <memory> |
| |
| #include <jni.h> |
| #include <SLES/OpenSLES.h> |
| |
| #include "webrtc/modules/audio_device/android/audio_common.h" |
| #include "webrtc/modules/audio_device/android/opensles_common.h" |
| #include "webrtc/modules/audio_device/audio_device_config.h" |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| #include "webrtc/modules/audio_device/include/audio_device_defines.h" |
| #include "webrtc/modules/utility/include/helpers_android.h" |
| #include "webrtc/modules/utility/include/jvm_android.h" |
| #include "webrtc/rtc_base/thread_checker.h" |
| |
| namespace webrtc { |
| |
| // Implements support for functions in the WebRTC audio stack for Android that |
| // relies on the AudioManager in android.media. It also populates an |
| // AudioParameter structure with native audio parameters detected at |
| // construction. This class does not make any audio-related modifications |
| // unless Init() is called. Caching audio parameters makes no changes but only |
| // reads data from the Java side. |
| class AudioManager { |
| public: |
| // Wraps the Java specific parts of the AudioManager into one helper class. |
| // Stores method IDs for all supported methods at construction and then |
| // allows calls like JavaAudioManager::Close() while hiding the Java/JNI |
| // parts that are associated with this call. |
| class JavaAudioManager { |
| public: |
| JavaAudioManager(NativeRegistration* native_registration, |
| std::unique_ptr<GlobalRef> audio_manager); |
| ~JavaAudioManager(); |
| |
| bool Init(); |
| void Close(); |
| bool IsCommunicationModeEnabled(); |
| bool IsDeviceBlacklistedForOpenSLESUsage(); |
| |
| private: |
| std::unique_ptr<GlobalRef> audio_manager_; |
| jmethodID init_; |
| jmethodID dispose_; |
| jmethodID is_communication_mode_enabled_; |
| jmethodID is_device_blacklisted_for_open_sles_usage_; |
| }; |
| |
| AudioManager(); |
| ~AudioManager(); |
| |
| // Sets the currently active audio layer combination. Must be called before |
| // Init(). |
| void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer); |
| |
| // Creates and realizes the main (global) Open SL engine object and returns |
| // a reference to it. The engine object is only created at the first call |
| // since OpenSL ES for Android only supports a single engine per application. |
| // Subsequent calls returns the already created engine. The SL engine object |
| // is destroyed when the AudioManager object is deleted. It means that the |
| // engine object will be the first OpenSL ES object to be created and last |
| // object to be destroyed. |
| // Note that NULL will be returned unless the audio layer is specified as |
| // AudioDeviceModule::kAndroidOpenSLESAudio or |
| // AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio. |
| SLObjectItf GetOpenSLEngine(); |
| |
| // Initializes the audio manager and stores the current audio mode. |
| bool Init(); |
| // Revert any setting done by Init(). |
| bool Close(); |
| |
| // Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION. |
| bool IsCommunicationModeEnabled() const; |
| |
| // Native audio parameters stored during construction. |
| const AudioParameters& GetPlayoutAudioParameters(); |
| const AudioParameters& GetRecordAudioParameters(); |
| |
| // Returns true if the device supports built-in audio effects for AEC, AGC |
| // and NS. Some devices can also be blacklisted for use in combination with |
| // platform effects and these devices will return false. |
| // Can currently only be used in combination with a Java based audio backend |
| // for the recoring side (i.e. using the android.media.AudioRecord API). |
| bool IsAcousticEchoCancelerSupported() const; |
| bool IsAutomaticGainControlSupported() const; |
| bool IsNoiseSuppressorSupported() const; |
| |
| // Returns true if the device supports the low-latency audio paths in |
| // combination with OpenSL ES. |
| bool IsLowLatencyPlayoutSupported() const; |
| bool IsLowLatencyRecordSupported() const; |
| |
| // Returns true if the device supports (and has been configured for) stereo. |
| // Call the Java API WebRtcAudioManager.setStereoOutput/Input() with true as |
| // paramter to enable stereo. Default is mono in both directions and the |
| // setting is set once and for all when the audio manager object is created. |
| // TODO(henrika): stereo is not supported in combination with OpenSL ES. |
| bool IsStereoPlayoutSupported() const; |
| bool IsStereoRecordSupported() const; |
| |
| // Returns true if the device supports pro-audio features in combination with |
| // OpenSL ES. |
| bool IsProAudioSupported() const; |
| |
| // Returns the estimated total delay of this device. Unit is in milliseconds. |
| // The vaule is set once at construction and never changes after that. |
| // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and |
| // webrtc::kHighLatencyModeDelayEstimateInMilliseconds. |
| int GetDelayEstimateInMilliseconds() const; |
| |
| private: |
| // Called from Java side so we can cache the native audio parameters. |
| // This method will be called by the WebRtcAudioManager constructor, i.e. |
| // on the same thread that this object is created on. |
| static void JNICALL 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, |
| jint output_buffer_size, |
| jint input_buffer_size, |
| jlong native_audio_manager); |
| void 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, |
| jint output_buffer_size, |
| jint input_buffer_size); |
| |
| // Stores thread ID in the constructor. |
| // We can then use ThreadChecker::CalledOnValidThread() to ensure that |
| // other methods are called from the same thread. |
| rtc::ThreadChecker thread_checker_; |
| |
| // Calls AttachCurrentThread() if this thread is not attached at construction. |
| // Also ensures that DetachCurrentThread() is called at destruction. |
| AttachCurrentThreadIfNeeded attach_thread_if_needed_; |
| |
| // Wraps the JNI interface pointer and methods associated with it. |
| std::unique_ptr<JNIEnvironment> j_environment_; |
| |
| // Contains factory method for creating the Java object. |
| std::unique_ptr<NativeRegistration> j_native_registration_; |
| |
| // Wraps the Java specific parts of the AudioManager. |
| std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_; |
| |
| // Contains the selected audio layer specified by the AudioLayer enumerator |
| // in the AudioDeviceModule class. |
| AudioDeviceModule::AudioLayer audio_layer_; |
| |
| // This object is the global entry point of the OpenSL ES API. |
| // After creating the engine object, the application can obtain this object‘s |
| // SLEngineItf interface. This interface contains creation methods for all |
| // the other object types in the API. None of these interface are realized |
| // by this class. It only provides access to the global engine object. |
| webrtc::ScopedSLObjectItf engine_object_; |
| |
| // Set to true by Init() and false by Close(). |
| bool initialized_; |
| |
| // True if device supports hardware (or built-in) AEC. |
| bool hardware_aec_; |
| // True if device supports hardware (or built-in) AGC. |
| bool hardware_agc_; |
| // True if device supports hardware (or built-in) NS. |
| bool hardware_ns_; |
| |
| // True if device supports the low-latency OpenSL ES audio path for output. |
| bool low_latency_playout_; |
| |
| // True if device supports the low-latency OpenSL ES audio path for input. |
| bool low_latency_record_; |
| |
| // True if device supports the low-latency OpenSL ES pro-audio path. |
| bool pro_audio_; |
| |
| // The delay estimate can take one of two fixed values depending on if the |
| // device supports low-latency output or not. |
| int delay_estimate_in_milliseconds_; |
| |
| // Contains native parameters (e.g. sample rate, channel configuration). |
| // Set at construction in OnCacheAudioParameters() which is called from |
| // Java on the same thread as this object is created on. |
| AudioParameters playout_parameters_; |
| AudioParameters record_parameters_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_ |