blob: 974ae90a61a36fa6137e57e45e6832417faf58fe [file] [log] [blame]
* 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.
#include "audio_device_generic.h"
#include "critical_section_wrapper.h"
#include <jni.h> // For accessing
#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
AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id);
// 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;
// Lock
void Lock()
void UnLock()
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
WebRtc_UWord32 _playQueueSeq;
// Recording buffer
WebRtc_Word8 _recQueueBuffer[N_REC_QUEUE_BUFFERS][2
WebRtc_UWord32 _recQueueSeq;
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