| /* |
| * 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 SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_ |
| #define SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_ |
| |
| #include <jni.h> |
| #include <stdio.h> |
| #include <stdlib.h> |
| |
| #include <SLES/OpenSLES.h> |
| #include <SLES/OpenSLES_Android.h> |
| #include <SLES/OpenSLES_AndroidConfiguration.h> |
| |
| #include <queue> |
| |
| #include "modules/audio_device/audio_device_generic.h" |
| #include "system_wrappers/interface/critical_section_wrapper.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; // Default fs |
| const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000; // Default fs |
| |
| const WebRtc_UWord32 N_REC_CHANNELS = 1; |
| const WebRtc_UWord32 N_PLAY_CHANNELS = 1; |
| |
| const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; |
| const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = 480; |
| |
| const WebRtc_UWord32 REC_MAX_TEMP_BUF_SIZE_PER_10ms = |
| N_REC_CHANNELS * REC_BUF_SIZE_IN_SAMPLES * sizeof(int16_t); |
| |
| const WebRtc_UWord32 PLAY_MAX_TEMP_BUF_SIZE_PER_10ms = |
| N_PLAY_CHANNELS * PLAY_BUF_SIZE_IN_SAMPLES * sizeof(int16_t); |
| |
| // Number of the buffers in playout queue |
| const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 8; |
| // Number of buffers in recording queue |
| // TODO(xian): Reduce the numbers of buffers to improve the latency. |
| const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 8; |
| // Some values returned from getMinBufferSize |
| // (Nexus S playout 72ms, recording 64ms) |
| // (Galaxy, 167ms, 44ms) |
| // (Nexus 7, 72ms, 48ms) |
| // (Xoom 92ms, 40ms) |
| |
| class ThreadWrapper; |
| |
| class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric { |
| public: |
| explicit AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id); |
| ~AudioDeviceAndroidOpenSLES(); |
| |
| // Retrieve the currently utilized audio layer |
| virtual WebRtc_Word32 |
| ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; // NOLINT |
| |
| // 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); // NOLINT |
| virtual WebRtc_Word32 InitPlayout(); |
| virtual bool PlayoutIsInitialized() const; |
| virtual WebRtc_Word32 RecordingIsAvailable(bool& available); // NOLINT |
| 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, // NOLINT |
| WebRtc_UWord16& volumeRight) const; // NOLINT |
| |
| // Audio mixer initialization |
| virtual WebRtc_Word32 SpeakerIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| virtual WebRtc_Word32 MicrophoneIsAvailable( |
| bool& available); |
| virtual WebRtc_Word32 InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| // Speaker volume controls |
| virtual WebRtc_Word32 SpeakerVolumeIsAvailable( |
| bool& available); // NOLINT |
| virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 SpeakerVolume( |
| WebRtc_UWord32& volume) const; // NOLINT |
| virtual WebRtc_Word32 MaxSpeakerVolume( |
| WebRtc_UWord32& maxVolume) const; // NOLINT |
| virtual WebRtc_Word32 MinSpeakerVolume( |
| WebRtc_UWord32& minVolume) const; // NOLINT |
| virtual WebRtc_Word32 SpeakerVolumeStepSize( |
| WebRtc_UWord16& stepSize) const; // NOLINT |
| |
| // Microphone volume controls |
| virtual WebRtc_Word32 MicrophoneVolumeIsAvailable( |
| bool& available); // NOLINT |
| virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 MicrophoneVolume( |
| WebRtc_UWord32& volume) const; // NOLINT |
| virtual WebRtc_Word32 MaxMicrophoneVolume( |
| WebRtc_UWord32& maxVolume) const; // NOLINT |
| virtual WebRtc_Word32 MinMicrophoneVolume( |
| WebRtc_UWord32& minVolume) const; // NOLINT |
| virtual WebRtc_Word32 |
| MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const; // NOLINT |
| |
| // Speaker mute control |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; // NOLINT |
| |
| // Microphone mute control |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const; // NOLINT |
| |
| // Microphone boost control |
| virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 SetMicrophoneBoost(bool enable); |
| virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const; // NOLINT |
| |
| // Stereo support |
| virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 SetStereoPlayout(bool enable); |
| virtual WebRtc_Word32 StereoPlayout(bool& enabled) const; // NOLINT |
| virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available); // NOLINT |
| virtual WebRtc_Word32 SetStereoRecording(bool enable); |
| virtual WebRtc_Word32 StereoRecording(bool& enabled) const; // NOLINT |
| |
| // Delay information and control |
| virtual WebRtc_Word32 |
| SetPlayoutBuffer(const AudioDeviceModule::BufferType type, |
| WebRtc_UWord16 sizeMS); |
| virtual WebRtc_Word32 PlayoutBuffer( |
| AudioDeviceModule::BufferType& type, // NOLINT |
| WebRtc_UWord16& sizeMS) const; |
| virtual WebRtc_Word32 PlayoutDelay( |
| WebRtc_UWord16& delayMS) const; // NOLINT |
| virtual WebRtc_Word32 RecordingDelay( |
| WebRtc_UWord16& delayMS) const; // NOLINT |
| |
| // CPU load |
| virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const; // NOLINT |
| |
| // 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; // NOLINT |
| |
| private: |
| // Lock |
| void Lock() { |
| crit_sect_.Enter(); |
| }; |
| void UnLock() { |
| crit_sect_.Leave(); |
| }; |
| |
| static void PlayerSimpleBufferQueueCallback( |
| SLAndroidSimpleBufferQueueItf queueItf, |
| void *pContext); |
| static void RecorderSimpleBufferQueueCallback( |
| SLAndroidSimpleBufferQueueItf queueItf, |
| void *pContext); |
| void PlayerSimpleBufferQueueCallbackHandler( |
| SLAndroidSimpleBufferQueueItf queueItf); |
| void RecorderSimpleBufferQueueCallbackHandler( |
| SLAndroidSimpleBufferQueueItf queueItf); |
| void CheckErr(SLresult res); |
| |
| // Delay updates |
| void UpdateRecordingDelay(); |
| void UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed); |
| |
| // Init |
| WebRtc_Word32 InitSampleRate(); |
| |
| // Misc |
| AudioDeviceBuffer* voe_audio_buffer_; |
| CriticalSectionWrapper& crit_sect_; |
| WebRtc_Word32 id_; |
| |
| // audio unit |
| SLObjectItf sles_engine_; |
| |
| // playout device |
| SLObjectItf sles_player_; |
| SLEngineItf sles_engine_itf_; |
| SLPlayItf sles_player_itf_; |
| SLAndroidSimpleBufferQueueItf sles_player_sbq_itf_; |
| SLObjectItf sles_output_mixer_; |
| SLVolumeItf sles_speaker_volume_; |
| |
| // recording device |
| SLObjectItf sles_recorder_; |
| SLRecordItf sles_recorder_itf_; |
| SLAndroidSimpleBufferQueueItf sles_recorder_sbq_itf_; |
| SLDeviceVolumeItf sles_mic_volume_; |
| WebRtc_UWord32 mic_dev_id_; |
| |
| WebRtc_UWord32 play_warning_, play_error_; |
| WebRtc_UWord32 rec_warning_, rec_error_; |
| |
| // States |
| bool is_recording_dev_specified_; |
| bool is_playout_dev_specified_; |
| bool is_initialized_; |
| bool is_recording_; |
| bool is_playing_; |
| bool is_rec_initialized_; |
| bool is_play_initialized_; |
| bool is_mic_initialized_; |
| bool is_speaker_initialized_; |
| |
| // Delay |
| WebRtc_UWord16 playout_delay_; |
| WebRtc_UWord16 recording_delay_; |
| |
| // AGC state |
| bool agc_enabled_; |
| |
| // Threads |
| ThreadWrapper* rec_thread_; |
| WebRtc_UWord32 rec_thread_id_; |
| static bool RecThreadFunc(void* context); |
| bool RecThreadFuncImpl(); |
| EventWrapper& rec_timer_; |
| |
| WebRtc_UWord32 mic_sampling_rate_; |
| WebRtc_UWord32 speaker_sampling_rate_; |
| WebRtc_UWord32 max_speaker_vol_; |
| WebRtc_UWord32 min_speaker_vol_; |
| bool loundspeaker_on_; |
| |
| SLDataFormat_PCM player_pcm_; |
| SLDataFormat_PCM record_pcm_; |
| |
| std::queue<WebRtc_Word8*> rec_queue_; |
| std::queue<WebRtc_Word8*> rec_voe_audio_queue_; |
| std::queue<WebRtc_Word8*> rec_voe_ready_queue_; |
| WebRtc_Word8 rec_buf_[N_REC_QUEUE_BUFFERS][ |
| N_REC_CHANNELS * sizeof(int16_t) * REC_BUF_SIZE_IN_SAMPLES]; |
| WebRtc_Word8 rec_voe_buf_[N_REC_QUEUE_BUFFERS][ |
| N_REC_CHANNELS * sizeof(int16_t) * REC_BUF_SIZE_IN_SAMPLES]; |
| |
| std::queue<WebRtc_Word8*> play_queue_; |
| WebRtc_Word8 play_buf_[N_PLAY_QUEUE_BUFFERS][ |
| N_PLAY_CHANNELS * sizeof(int16_t) * PLAY_BUF_SIZE_IN_SAMPLES]; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_ |
