| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H |
| |
| #include "audio_device_generic.h" |
| #include "critical_section_wrapper.h" |
| |
| #include <jni.h> // For accessing AudioDeviceAndroid.java |
| #include <stdio.h> |
| #include <stdlib.h> |
| |
| #include <SLES/OpenSLES.h> |
| #include <SLES/OpenSLES_Android.h> |
| #include <SLES/OpenSLES_AndroidConfiguration.h> |
| |
| namespace webrtc |
| { |
| class EventWrapper; |
| |
| const WebRtc_UWord32 N_MAX_INTERFACES = 3; |
| const WebRtc_UWord32 N_MAX_OUTPUT_DEVICES = 6; |
| const WebRtc_UWord32 N_MAX_INPUT_DEVICES = 3; |
| |
| const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 16000;//44000; // Default fs |
| const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000;//44000; // Default fs |
| |
| 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 |
| const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = 480; |
| |
| // Number of the buffers in playout queue |
| const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 2; |
| // Number of buffers in recording queue |
| const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 2; |
| // Number of 10 ms recording blocks in rec buffer |
| const WebRtc_UWord16 N_REC_BUFFERS = 20; |
| |
| class ThreadWrapper; |
| |
| class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric |
| { |
| public: |
| AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id); |
| ~AudioDeviceAndroidOpenSLES(); |
| |
| // Retrieve the currently utilized audio layer |
| virtual WebRtc_Word32 |
| ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; |
| |
| // Main initializaton and termination |
| virtual WebRtc_Word32 Init(); |
| virtual WebRtc_Word32 Terminate(); |
| virtual bool Initialized() const; |
| |
| // Device enumeration |
| virtual WebRtc_Word16 PlayoutDevices(); |
| virtual WebRtc_Word16 RecordingDevices(); |
| virtual WebRtc_Word32 |
| PlayoutDeviceName(WebRtc_UWord16 index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]); |
| virtual WebRtc_Word32 |
| RecordingDeviceName(WebRtc_UWord16 index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]); |
| |
| // Device selection |
| 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); |
| |
| // Audio transport initialization |
| 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; |
| |
| // Audio transport control |
| virtual WebRtc_Word32 StartPlayout(); |
| virtual WebRtc_Word32 StopPlayout(); |
| virtual bool Playing() const; |
| virtual WebRtc_Word32 StartRecording(); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual bool Recording() const; |
| |
| // Microphone Automatic Gain Control (AGC) |
| virtual WebRtc_Word32 SetAGC(bool enable); |
| virtual bool AGC() const; |
| |
| // Volume control based on the Windows Wave API (Windows only) |
| virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, |
| WebRtc_UWord16 volumeRight); |
| virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, |
| WebRtc_UWord16& volumeRight) const; |
| |
| // Audio mixer initialization |
| virtual WebRtc_Word32 SpeakerIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| SLPlayItf playItf; |
| virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| // Speaker volume controls |
| 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; |
| |
| // Microphone volume controls |
| 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; |
| |
| // Speaker mute control |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; |
| |
| // Microphone mute control |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const; |
| |
| // Microphone boost control |
| virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneBoost(bool enable); |
| virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const; |
| |
| // Stereo support |
| 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; |
| |
| // Delay information and control |
| 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; |
| |
| // CPU load |
| virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const; |
| |
| // Error and warning information |
| 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(); |
| |
| // Attach audio buffer |
| virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| // Speaker audio routing |
| virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable); |
| virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const; |
| |
| private: |
| // Lock |
| void Lock() |
| { |
| _critSect.Enter(); |
| }; |
| void UnLock() |
| { |
| _critSect.Leave(); |
| }; |
| |
| static void PlayerSimpleBufferQueueCallback( |
| SLAndroidSimpleBufferQueueItf queueItf, |
| void *pContext); |
| void PlayerSimpleBufferQueueCallbackHandler( |
| SLAndroidSimpleBufferQueueItf queueItf); |
| static void RecorderSimpleBufferQueueCallback( |
| SLAndroidSimpleBufferQueueItf queueItf, |
| void *pContext); |
| void RecorderSimpleBufferQueueCallbackHandler( |
| SLAndroidSimpleBufferQueueItf queueItf); |
| void CheckErr(SLresult res); |
| |
| // Delay updates |
| void UpdateRecordingDelay(); |
| void UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed); |
| |
| // Init |
| WebRtc_Word32 InitSampleRate(); |
| |
| // Threads |
| static bool RecThreadFunc(void*); |
| static bool PlayThreadFunc(void*); |
| bool RecThreadProcess(); |
| bool PlayThreadProcess(); |
| |
| // Misc |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| WebRtc_Word32 _id; |
| |
| // audio unit |
| SLObjectItf _slEngineObject; |
| |
| // playout device |
| SLObjectItf _slPlayer; |
| SLEngineItf _slEngine; |
| SLPlayItf _slPlayerPlay; |
| SLAndroidSimpleBufferQueueItf _slPlayerSimpleBufferQueue; |
| SLObjectItf _slOutputMixObject; |
| SLVolumeItf _slSpeakerVolume; |
| |
| // recording device |
| SLObjectItf _slRecorder; |
| SLRecordItf _slRecorderRecord; |
| SLAudioIODeviceCapabilitiesItf _slAudioIODeviceCapabilities; |
| SLAndroidSimpleBufferQueueItf _slRecorderSimpleBufferQueue; |
| SLDeviceVolumeItf _slMicVolume; |
| |
| WebRtc_UWord32 _micDeviceId; |
| |
| // Events |
| EventWrapper& _timeEventRec; |
| // Threads |
| ThreadWrapper* _ptrThreadRec; |
| WebRtc_UWord32 _recThreadID; |
| // TODO(xians), remove the following flag |
| bool _recThreadIsInitialized; |
| |
| // Playout buffer |
| WebRtc_Word8 _playQueueBuffer[N_PLAY_QUEUE_BUFFERS][2 |
| * PLAY_BUF_SIZE_IN_SAMPLES]; |
| WebRtc_UWord32 _playQueueSeq; |
| // Recording buffer |
| WebRtc_Word8 _recQueueBuffer[N_REC_QUEUE_BUFFERS][2 |
| * REC_BUF_SIZE_IN_SAMPLES]; |
| WebRtc_UWord32 _recQueueSeq; |
| WebRtc_Word8 _recBuffer[N_REC_BUFFERS][2*REC_BUF_SIZE_IN_SAMPLES]; |
| WebRtc_UWord32 _recLength[N_REC_BUFFERS]; |
| WebRtc_UWord32 _recSeqNumber[N_REC_BUFFERS]; |
| WebRtc_UWord32 _recCurrentSeq; |
| // Current total size all data in buffers, used for delay estimate |
| WebRtc_UWord32 _recBufferTotalSize; |
| |
| // States |
| bool _recordingDeviceIsSpecified; |
| bool _playoutDeviceIsSpecified; |
| bool _initialized; |
| bool _recording; |
| bool _playing; |
| bool _recIsInitialized; |
| bool _playIsInitialized; |
| bool _micIsInitialized; |
| bool _speakerIsInitialized; |
| |
| // Warnings and errors |
| WebRtc_UWord16 _playWarning; |
| WebRtc_UWord16 _playError; |
| WebRtc_UWord16 _recWarning; |
| WebRtc_UWord16 _recError; |
| |
| // Delay |
| WebRtc_UWord16 _playoutDelay; |
| WebRtc_UWord16 _recordingDelay; |
| |
| // AGC state |
| bool _AGC; |
| |
| // The sampling rate to use with Audio Device Buffer |
| WebRtc_UWord32 _adbSampleRate; |
| // Stored device properties |
| WebRtc_UWord32 _samplingRateIn; // Sampling frequency for Mic |
| WebRtc_UWord32 _samplingRateOut; // Sampling frequency for Speaker |
| WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value |
| WebRtc_UWord32 _minSpeakerVolume; // The minimum speaker volume value |
| bool _loudSpeakerOn; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H |
