| /* |
| * Copyright (c) 2011 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_ANDROID_JNI_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H |
| |
| #include "audio_device_generic.h" |
| #include "critical_section_wrapper.h" |
| |
| #include <jni.h> // For accessing AudioDeviceAndroid java class |
| |
| namespace webrtc |
| { |
| class EventWrapper; |
| |
| const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz |
| const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz |
| |
| const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording |
| const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout |
| |
| const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz |
| |
| |
| WebRtc_Word32 SetAndroidAudioDeviceObjects(void* javaVM, void* env, |
| void* context); |
| |
| class ThreadWrapper; |
| |
| class AudioDeviceAndroidJni: public AudioDeviceGeneric |
| { |
| public: |
| AudioDeviceAndroidJni(const WebRtc_Word32 id); |
| ~AudioDeviceAndroidJni(); |
| |
| virtual WebRtc_Word32 ActiveAudioLayer( |
| AudioDeviceModule::AudioLayer& audioLayer) const; |
| |
| virtual WebRtc_Word32 Init(); |
| virtual WebRtc_Word32 Terminate(); |
| virtual bool Initialized() const; |
| |
| virtual WebRtc_Word16 PlayoutDevices(); |
| virtual WebRtc_Word16 RecordingDevices(); |
| virtual WebRtc_Word32 PlayoutDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| virtual WebRtc_Word32 RecordingDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| |
| virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| |
| virtual WebRtc_Word32 PlayoutIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitPlayout(); |
| virtual bool PlayoutIsInitialized() const; |
| virtual WebRtc_Word32 RecordingIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitRecording(); |
| virtual bool RecordingIsInitialized() const; |
| |
| virtual WebRtc_Word32 StartPlayout(); |
| virtual WebRtc_Word32 StopPlayout(); |
| virtual bool Playing() const; |
| virtual WebRtc_Word32 StartRecording(); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual bool Recording() const; |
| |
| virtual WebRtc_Word32 SetAGC(bool enable); |
| virtual bool AGC() const; |
| |
| virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, |
| WebRtc_UWord16 volumeRight); |
| virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, |
| WebRtc_UWord16& volumeRight) const; |
| |
| virtual WebRtc_Word32 SpeakerIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const; |
| virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const; |
| virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const; |
| virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const; |
| |
| virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const; |
| virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const; |
| virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const; |
| virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) |
| const; |
| |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; |
| |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const; |
| |
| virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneBoost(bool enable); |
| virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const; |
| |
| virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetStereoPlayout(bool enable); |
| virtual WebRtc_Word32 StereoPlayout(bool& enabled) const; |
| virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetStereoRecording(bool enable); |
| virtual WebRtc_Word32 StereoRecording(bool& enabled) const; |
| |
| virtual WebRtc_Word32 SetPlayoutBuffer( |
| const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS); |
| virtual WebRtc_Word32 PlayoutBuffer( |
| AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const; |
| virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const; |
| virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const; |
| |
| virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& 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 WebRtc_Word32 SetRecordingSampleRate( |
| const WebRtc_UWord32 samplesPerSec); |
| virtual WebRtc_Word32 SetPlayoutSampleRate( |
| const WebRtc_UWord32 samplesPerSec); |
| |
| virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable); |
| virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const; |
| |
| private: |
| // Lock |
| void Lock() |
| { |
| _critSect.Enter(); |
| }; |
| void UnLock() |
| { |
| _critSect.Leave(); |
| }; |
| |
| // Init |
| WebRtc_Word32 InitJavaResources(); |
| WebRtc_Word32 InitSampleRate(); |
| |
| // Threads |
| static bool RecThreadFunc(void*); |
| static bool PlayThreadFunc(void*); |
| bool RecThreadProcess(); |
| bool PlayThreadProcess(); |
| |
| // Misc |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| WebRtc_Word32 _id; |
| |
| // Events |
| EventWrapper& _timeEventRec; |
| EventWrapper& _timeEventPlay; |
| EventWrapper& _recStartStopEvent; |
| EventWrapper& _playStartStopEvent; |
| |
| // Threads |
| ThreadWrapper* _ptrThreadPlay; |
| ThreadWrapper* _ptrThreadRec; |
| WebRtc_UWord32 _recThreadID; |
| WebRtc_UWord32 _playThreadID; |
| bool _playThreadIsInitialized; |
| bool _recThreadIsInitialized; |
| bool _shutdownPlayThread; |
| bool _shutdownRecThread; |
| |
| // Rec buffer |
| WebRtc_Word8 _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 |
| WebRtc_UWord16 _playWarning; |
| WebRtc_UWord16 _playError; |
| WebRtc_UWord16 _recWarning; |
| WebRtc_UWord16 _recError; |
| |
| // Delay |
| WebRtc_UWord16 _delayPlayout; |
| WebRtc_UWord16 _delayRecording; |
| |
| // AGC state |
| bool _AGC; |
| |
| // Stored device properties |
| WebRtc_UWord16 _samplingFreqIn; // Sampling frequency for Mic |
| WebRtc_UWord16 _samplingFreqOut; // Sampling frequency for Speaker |
| WebRtc_UWord32 _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 |
| jobject _javaContext; // the application context |
| |
| 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 |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H |
