| /* |
| * 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_AUDIO_RECORD_JNI_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_ |
| |
| #include <jni.h> |
| |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/modules/audio_device/include/audio_device_defines.h" |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| |
| namespace webrtc { |
| |
| class EventWrapper; |
| class ThreadWrapper; |
| class PlayoutDelayProvider; |
| |
| const uint32_t N_REC_SAMPLES_PER_SEC = 16000; // Default is 16 kHz |
| const uint32_t N_REC_CHANNELS = 1; // default is mono recording |
| const uint32_t REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz |
| |
| class AudioRecordJni { |
| public: |
| static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* env, |
| void* context); |
| static void ClearAndroidAudioDeviceObjects(); |
| |
| AudioRecordJni(const int32_t id, PlayoutDelayProvider* delay_provider); |
| ~AudioRecordJni(); |
| |
| // Main initializaton and termination |
| int32_t Init(); |
| int32_t Terminate(); |
| bool Initialized() const { return _initialized; } |
| |
| // Device enumeration |
| int16_t RecordingDevices() { return 1; } // There is one device only |
| 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); |
| |
| // Audio transport initialization |
| int32_t RecordingIsAvailable(bool& available); // NOLINT |
| int32_t InitRecording(); |
| bool RecordingIsInitialized() const { return _recIsInitialized; } |
| |
| // 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; } |
| |
| // Audio mixer initialization |
| int32_t InitMicrophone(); |
| bool MicrophoneIsInitialized() const { return _micIsInitialized; } |
| |
| // 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); |
| int32_t MicrophoneVolume(uint32_t& volume) const; // NOLINT |
| int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const; // 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); |
| int32_t MicrophoneMute(bool& enabled) const; // 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); |
| int32_t StereoRecording(bool& enabled) const; // NOLINT |
| |
| // Delay information and control |
| int32_t RecordingDelay(uint16_t& delayMS) const; // NOLINT |
| |
| bool RecordingWarning() const; |
| bool RecordingError() const; |
| void ClearRecordingWarning(); |
| void ClearRecordingError(); |
| |
| // Attach audio buffer |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| int32_t SetRecordingSampleRate(const uint32_t samplesPerSec); |
| |
| bool BuiltInAECIsAvailable() const; |
| int32_t EnableBuiltInAEC(bool enable); |
| |
| private: |
| void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { |
| _critSect.Enter(); |
| } |
| void UnLock() UNLOCK_FUNCTION(_critSect) { |
| _critSect.Leave(); |
| } |
| |
| int32_t InitJavaResources(); |
| int32_t InitSampleRate(); |
| |
| static bool RecThreadFunc(void*); |
| bool RecThreadProcess(); |
| |
| // TODO(leozwang): Android holds only one JVM, all these jni handling |
| // will be consolidated into a single place to make it consistant and |
| // reliable. Chromium has a good example at base/android. |
| static JavaVM* globalJvm; |
| static JNIEnv* globalJNIEnv; |
| static jobject globalContext; |
| static jclass globalScClass; |
| |
| JavaVM* _javaVM; // denotes a Java VM |
| JNIEnv* _jniEnvRec; // The JNI env for recording thread |
| jclass _javaScClass; // AudioDeviceAndroid class |
| jobject _javaScObj; // AudioDeviceAndroid object |
| jobject _javaRecBuffer; |
| void* _javaDirectRecBuffer; // Direct buffer pointer to rec buffer |
| jmethodID _javaMidRecAudio; // Method ID of rec in AudioDeviceAndroid |
| |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| int32_t _id; |
| PlayoutDelayProvider* _delay_provider; |
| bool _initialized; |
| |
| EventWrapper& _timeEventRec; |
| EventWrapper& _recStartStopEvent; |
| ThreadWrapper* _ptrThreadRec; |
| uint32_t _recThreadID; |
| bool _recThreadIsInitialized; |
| bool _shutdownRecThread; |
| |
| int8_t _recBuffer[2 * REC_BUF_SIZE_IN_SAMPLES]; |
| bool _recordingDeviceIsSpecified; |
| |
| bool _recording; |
| bool _recIsInitialized; |
| bool _micIsInitialized; |
| |
| bool _startRec; |
| |
| uint16_t _recWarning; |
| uint16_t _recError; |
| |
| uint16_t _delayRecording; |
| |
| bool _AGC; |
| |
| uint16_t _samplingFreqIn; // Sampling frequency for Mic |
| int _recAudioSource; |
| |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_ |