/* * Copyright (c) 2013 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_OPENSLES_INPUT_H_ #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_INPUT_H_ #include #include #include #include "webrtc/modules/audio_device/android/audio_manager_jni.h" #include "webrtc/modules/audio_device/android/low_latency_event.h" #include "webrtc/modules/audio_device/include/audio_device.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { class AudioDeviceBuffer; class CriticalSectionWrapper; class PlayoutDelayProvider; class SingleRwFifo; class ThreadWrapper; // OpenSL implementation that facilitate capturing PCM data from an android // device's microphone. // This class is Thread-compatible. I.e. Given an instance of this class, calls // to non-const methods require exclusive access to the object. class OpenSlesInput { public: OpenSlesInput(const int32_t id, PlayoutDelayProvider* delay_provider); ~OpenSlesInput(); static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* env, void* context); static void ClearAndroidAudioDeviceObjects(); // Main initializaton and termination int32_t Init(); int32_t Terminate(); bool Initialized() const { return initialized_; } // Device enumeration int16_t RecordingDevices() { return 1; } int32_t RecordingDeviceName(uint16_t index, char name[kAdmMaxDeviceNameSize], char guid[kAdmMaxGuidSize]); // Device selection int32_t SetRecordingDevice(uint16_t index); int32_t SetRecordingDevice( AudioDeviceModule::WindowsDeviceType device) { return -1; } // No-op int32_t SetRecordingSampleRate(uint32_t sample_rate_hz) { return 0; } // Audio transport initialization int32_t RecordingIsAvailable(bool& available); // NOLINT int32_t InitRecording(); bool RecordingIsInitialized() const { return rec_initialized_; } // Audio transport control int32_t StartRecording(); int32_t StopRecording(); bool Recording() const { return recording_; } // Microphone Automatic Gain Control (AGC) int32_t SetAGC(bool enable); bool AGC() const { return agc_enabled_; } // Audio mixer initialization int32_t MicrophoneIsAvailable(bool& available); // NOLINT int32_t InitMicrophone(); bool MicrophoneIsInitialized() const { return mic_initialized_; } // Microphone volume controls int32_t MicrophoneVolumeIsAvailable(bool& available); // NOLINT // TODO(leozwang): Add microphone volume control when OpenSL APIs // are available. int32_t SetMicrophoneVolume(uint32_t volume) { return 0; } int32_t MicrophoneVolume(uint32_t& volume) const { return -1; } // NOLINT int32_t MaxMicrophoneVolume( uint32_t& maxVolume) const { return 0; } // NOLINT int32_t MinMicrophoneVolume(uint32_t& minVolume) const; // NOLINT int32_t MicrophoneVolumeStepSize( uint16_t& stepSize) const; // NOLINT // Microphone mute control int32_t MicrophoneMuteIsAvailable(bool& available); // NOLINT int32_t SetMicrophoneMute(bool enable) { return -1; } int32_t MicrophoneMute(bool& enabled) const { return -1; } // NOLINT // Microphone boost control int32_t MicrophoneBoostIsAvailable(bool& available); // NOLINT int32_t SetMicrophoneBoost(bool enable); int32_t MicrophoneBoost(bool& enabled) const; // NOLINT // Stereo support int32_t StereoRecordingIsAvailable(bool& available); // NOLINT int32_t SetStereoRecording(bool enable) { return -1; } int32_t StereoRecording(bool& enabled) const; // NOLINT // Delay information and control int32_t RecordingDelay(uint16_t& delayMS) const; // NOLINT bool RecordingWarning() const { return false; } bool RecordingError() const { return false; } void ClearRecordingWarning() {} void ClearRecordingError() {} // Attach audio buffer void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); private: enum { kNumInterfaces = 2, // Keep as few OpenSL buffers as possible to avoid wasting memory. 2 is // minimum for playout. Keep 2 for recording as well. kNumOpenSlBuffers = 2, kNum10MsToBuffer = 3, }; int InitSampleRate(); int buffer_size_samples() const; int buffer_size_bytes() const; void UpdateRecordingDelay(); void UpdateSampleRate(); void CalculateNumFifoBuffersNeeded(); void AllocateBuffers(); int TotalBuffersUsed() const; bool EnqueueAllBuffers(); // This function also configures the audio recorder, e.g. sample rate to use // etc, so it should be called when starting recording. bool CreateAudioRecorder(); void DestroyAudioRecorder(); // When overrun happens there will be more frames received from OpenSL than // the desired number of buffers. It is possible to expand the number of // buffers as you go but that would greatly increase the complexity of this // code. HandleOverrun gracefully handles the scenario by restarting playout, // throwing away all pending audio data. This will sound like a click. This // is also logged to identify these types of clicks. // This function returns true if there has been overrun. Further processing // of audio data should be avoided until this function returns false again. // The function needs to be protected by |crit_sect_|. bool HandleOverrun(int event_id, int event_msg); static void RecorderSimpleBufferQueueCallback( SLAndroidSimpleBufferQueueItf queueItf, void* pContext); // This function must not take any locks or do any heavy work. It is a // requirement for the OpenSL implementation to work as intended. The reason // for this is that taking locks exposes the OpenSL thread to the risk of // priority inversion. void RecorderSimpleBufferQueueCallbackHandler( SLAndroidSimpleBufferQueueItf queueItf); bool StartCbThreads(); void StopCbThreads(); static bool CbThread(void* context); // This function must be protected against data race with threads calling this // class' public functions. It is a requirement for this class to be // Thread-compatible. bool CbThreadImpl(); // Java API handle AudioManagerJni audio_manager_; int id_; PlayoutDelayProvider* delay_provider_; bool initialized_; bool mic_initialized_; bool rec_initialized_; // Members that are read/write accessed concurrently by the process thread and // threads calling public functions of this class. scoped_ptr rec_thread_; // Processing thread scoped_ptr crit_sect_; // This member controls the starting and stopping of recording audio to the // the device. bool recording_; // Only one thread, T1, may push and only one thread, T2, may pull. T1 may or // may not be the same thread as T2. T2 is the process thread and T1 is the // OpenSL thread. scoped_ptr fifo_; int num_fifo_buffers_needed_; LowLatencyEvent event_; int number_overruns_; // OpenSL handles SLObjectItf sles_engine_; SLEngineItf sles_engine_itf_; SLObjectItf sles_recorder_; SLRecordItf sles_recorder_itf_; SLAndroidSimpleBufferQueueItf sles_recorder_sbq_itf_; // Audio buffers AudioDeviceBuffer* audio_buffer_; // Holds all allocated memory such that it is deallocated properly. scoped_array > rec_buf_; // Index in |rec_buf_| pointing to the audio buffer that will be ready the // next time RecorderSimpleBufferQueueCallbackHandler is invoked. // Ready means buffer contains audio data from the device. int active_queue_; // Audio settings uint32_t rec_sampling_rate_; bool agc_enabled_; // Audio status uint16_t recording_delay_; }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_INPUT_H_