| /* |
| * 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. |
| */ |
| |
| /* |
| * Android audio device interface (JNI/AudioTrack/AudioRecord usage) |
| */ |
| |
| #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_JNI_ANDROID_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_JNI_ANDROID_H |
| |
| #include "audio_device_generic.h" |
| #include "critical_section_wrapper.h" |
| |
| #include <jni.h> // For accessing AudioDeviceAndroid java class |
| |
| namespace webrtc |
| { |
| class EventWrapper; |
| |
| const uint32_t N_REC_SAMPLES_PER_SEC = 16000; // Default is 16 kHz |
| const uint32_t N_PLAY_SAMPLES_PER_SEC = 16000; // Default is 16 kHz |
| |
| const uint32_t N_REC_CHANNELS = 1; // default is mono recording |
| const uint32_t N_PLAY_CHANNELS = 1; // default is mono playout |
| |
| const uint32_t REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz |
| |
| |
| class ThreadWrapper; |
| |
| class AudioDeviceAndroidJni : public AudioDeviceGeneric { |
| public: |
| AudioDeviceAndroidJni(const int32_t id); |
| ~AudioDeviceAndroidJni(); |
| |
| static int32_t SetAndroidAudioDeviceObjects(void* javaVM, |
| void* env, |
| void* context); |
| |
| virtual int32_t ActiveAudioLayer( |
| AudioDeviceModule::AudioLayer& audioLayer) const; |
| |
| virtual int32_t Init(); |
| virtual int32_t Terminate(); |
| virtual bool Initialized() const; |
| |
| virtual int16_t PlayoutDevices(); |
| virtual int16_t RecordingDevices(); |
| virtual int32_t PlayoutDeviceName(uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]); |
| virtual int32_t RecordingDeviceName( |
| uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]); |
| |
| virtual int32_t SetPlayoutDevice(uint16_t index); |
| virtual int32_t SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| virtual int32_t SetRecordingDevice(uint16_t index); |
| virtual int32_t SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| |
| virtual int32_t PlayoutIsAvailable(bool& available); |
| virtual int32_t InitPlayout(); |
| virtual bool PlayoutIsInitialized() const; |
| virtual int32_t RecordingIsAvailable(bool& available); |
| virtual int32_t InitRecording(); |
| virtual bool RecordingIsInitialized() const; |
| |
| virtual int32_t StartPlayout(); |
| virtual int32_t StopPlayout(); |
| virtual bool Playing() const; |
| virtual int32_t StartRecording(); |
| virtual int32_t StopRecording(); |
| virtual bool Recording() const; |
| |
| virtual int32_t SetAGC(bool enable); |
| virtual bool AGC() const; |
| |
| virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight); |
| virtual int32_t WaveOutVolume(uint16_t& volumeLeft, |
| uint16_t& volumeRight) const; |
| |
| virtual int32_t SpeakerIsAvailable(bool& available); |
| virtual int32_t InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| virtual int32_t MicrophoneIsAvailable(bool& available); |
| virtual int32_t InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| virtual int32_t SpeakerVolumeIsAvailable(bool& available); |
| virtual int32_t SetSpeakerVolume(uint32_t volume); |
| virtual int32_t SpeakerVolume(uint32_t& volume) const; |
| virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const; |
| virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const; |
| virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const; |
| |
| virtual int32_t MicrophoneVolumeIsAvailable(bool& available); |
| virtual int32_t SetMicrophoneVolume(uint32_t volume); |
| virtual int32_t MicrophoneVolume(uint32_t& volume) const; |
| virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const; |
| virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const; |
| virtual int32_t MicrophoneVolumeStepSize( |
| uint16_t& stepSize) const; |
| |
| virtual int32_t SpeakerMuteIsAvailable(bool& available); |
| virtual int32_t SetSpeakerMute(bool enable); |
| virtual int32_t SpeakerMute(bool& enabled) const; |
| |
| virtual int32_t MicrophoneMuteIsAvailable(bool& available); |
| virtual int32_t SetMicrophoneMute(bool enable); |
| virtual int32_t MicrophoneMute(bool& enabled) const; |
| |
| virtual int32_t MicrophoneBoostIsAvailable(bool& available); |
| virtual int32_t SetMicrophoneBoost(bool enable); |
| virtual int32_t MicrophoneBoost(bool& enabled) const; |
| |
| virtual int32_t StereoPlayoutIsAvailable(bool& available); |
| virtual int32_t SetStereoPlayout(bool enable); |
| virtual int32_t StereoPlayout(bool& enabled) const; |
| virtual int32_t StereoRecordingIsAvailable(bool& available); |
| virtual int32_t SetStereoRecording(bool enable); |
| virtual int32_t StereoRecording(bool& enabled) const; |
| |
| virtual int32_t SetPlayoutBuffer( |
| const AudioDeviceModule::BufferType type, uint16_t sizeMS); |
| virtual int32_t PlayoutBuffer( |
| AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const; |
| virtual int32_t PlayoutDelay(uint16_t& delayMS) const; |
| virtual int32_t RecordingDelay(uint16_t& delayMS) const; |
| |
| virtual int32_t CPULoad(uint16_t& load) const; |
| |
| virtual bool PlayoutWarning() const; |
| virtual bool PlayoutError() const; |
| virtual bool RecordingWarning() const; |
| virtual bool RecordingError() const; |
| virtual void ClearPlayoutWarning(); |
| virtual void ClearPlayoutError(); |
| virtual void ClearRecordingWarning(); |
| virtual void ClearRecordingError(); |
| |
| virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| virtual int32_t SetRecordingSampleRate( |
| const uint32_t samplesPerSec); |
| virtual int32_t SetPlayoutSampleRate( |
| const uint32_t samplesPerSec); |
| |
| virtual int32_t SetLoudspeakerStatus(bool enable); |
| virtual int32_t GetLoudspeakerStatus(bool& enable) const; |
| |
| private: |
| // Lock |
| void Lock() { |
| _critSect.Enter(); |
| }; |
| void UnLock() { |
| _critSect.Leave(); |
| }; |
| |
| // Init |
| int32_t InitJavaResources(); |
| int32_t InitSampleRate(); |
| |
| // Threads |
| static bool RecThreadFunc(void*); |
| static bool PlayThreadFunc(void*); |
| bool RecThreadProcess(); |
| bool PlayThreadProcess(); |
| |
| // Misc |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| int32_t _id; |
| |
| // Events |
| EventWrapper& _timeEventRec; |
| EventWrapper& _timeEventPlay; |
| EventWrapper& _recStartStopEvent; |
| EventWrapper& _playStartStopEvent; |
| |
| // Threads |
| ThreadWrapper* _ptrThreadPlay; |
| ThreadWrapper* _ptrThreadRec; |
| uint32_t _recThreadID; |
| uint32_t _playThreadID; |
| bool _playThreadIsInitialized; |
| bool _recThreadIsInitialized; |
| bool _shutdownPlayThread; |
| bool _shutdownRecThread; |
| |
| // Rec buffer |
| int8_t _recBuffer[2 * REC_BUF_SIZE_IN_SAMPLES]; |
| |
| // States |
| bool _recordingDeviceIsSpecified; |
| bool _playoutDeviceIsSpecified; |
| bool _initialized; |
| bool _recording; |
| bool _playing; |
| bool _recIsInitialized; |
| bool _playIsInitialized; |
| bool _micIsInitialized; |
| bool _speakerIsInitialized; |
| |
| // Signal flags to threads |
| bool _startRec; |
| bool _stopRec; |
| bool _startPlay; |
| bool _stopPlay; |
| |
| // Warnings and errors |
| uint16_t _playWarning; |
| uint16_t _playError; |
| uint16_t _recWarning; |
| uint16_t _recError; |
| |
| // Delay |
| uint16_t _delayPlayout; |
| uint16_t _delayRecording; |
| |
| // AGC state |
| bool _AGC; |
| |
| // Stored device properties |
| uint16_t _samplingFreqIn; // Sampling frequency for Mic |
| uint16_t _samplingFreqOut; // Sampling frequency for Speaker |
| uint32_t _maxSpeakerVolume; // The maximum speaker volume value |
| bool _loudSpeakerOn; |
| // Stores the desired audio source to use, set in SetRecordingDevice |
| int _recAudioSource; |
| |
| // JNI and Java |
| JavaVM* _javaVM; // denotes a Java VM |
| |
| JNIEnv* _jniEnvPlay; // The JNI env for playout thread |
| JNIEnv* _jniEnvRec; // The JNI env for recording thread |
| |
| jclass _javaScClass; // AudioDeviceAndroid class |
| jobject _javaScObj; // AudioDeviceAndroid object |
| |
| // The play buffer field in AudioDeviceAndroid object (global ref) |
| jobject _javaPlayBuffer; |
| // The rec buffer field in AudioDeviceAndroid object (global ref) |
| jobject _javaRecBuffer; |
| void* _javaDirectPlayBuffer; // Direct buffer pointer to play buffer |
| void* _javaDirectRecBuffer; // Direct buffer pointer to rec buffer |
| jmethodID _javaMidPlayAudio; // Method ID of play in AudioDeviceAndroid |
| jmethodID _javaMidRecAudio; // Method ID of rec in AudioDeviceAndroid |
| |
| // 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; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_JNI_ANDROID_H |