| /* |
| ** Copyright 2008, The Android Open-Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| #ifndef ANDROID_AUDIO_HARDWARE_H |
| #define ANDROID_AUDIO_HARDWARE_H |
| |
| #include <stdint.h> |
| #include <sys/types.h> |
| |
| #include <utils/threads.h> |
| #include <utils/SortedVector.h> |
| |
| #include <hardware_legacy/AudioHardwareBase.h> |
| #include <media/mediarecorder.h> |
| |
| #include "secril-client.h" |
| |
| extern "C" { |
| struct pcm; |
| struct mixer; |
| struct mixer_ctl; |
| }; |
| |
| namespace android { |
| |
| // TODO: determine actual audio DSP and hardware latency |
| // Additionnal latency introduced by audio DSP and hardware in ms |
| #define AUDIO_HW_OUT_LATENCY_MS 0 |
| // Default audio output sample rate |
| #define AUDIO_HW_OUT_SAMPLERATE 44100 |
| // Default audio output channel mask |
| #define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO) |
| // Default audio output sample format |
| #define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT) |
| // Kernel pcm out buffer size in frames at 44.1kHz |
| #define AUDIO_HW_OUT_PERIOD_MULT 8 // (8 * 128 = 1024 frames) |
| #define AUDIO_HW_OUT_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_OUT_PERIOD_MULT) |
| #define AUDIO_HW_OUT_PERIOD_CNT 4 |
| // Default audio output buffer size in bytes |
| #define AUDIO_HW_OUT_PERIOD_BYTES (AUDIO_HW_OUT_PERIOD_SZ * 2 * sizeof(int16_t)) |
| |
| // Default audio input sample rate |
| #define AUDIO_HW_IN_SAMPLERATE 8000 |
| // Default audio input channel mask |
| #define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_MONO) |
| // Default audio input sample format |
| #define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT) |
| // Number of buffers in audio driver for input |
| #define AUDIO_HW_NUM_IN_BUF 2 |
| // Kernel pcm in buffer size in frames at 44.1kHz (before resampling) |
| #define AUDIO_HW_IN_PERIOD_MULT 16 // (16 * 128 = 2048 frames) |
| #define AUDIO_HW_IN_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_IN_PERIOD_MULT) |
| #define AUDIO_HW_IN_PERIOD_CNT 2 |
| // Default audio input buffer size in bytes (8kHz mono) |
| #define AUDIO_HW_IN_PERIOD_BYTES ((AUDIO_HW_IN_PERIOD_SZ*sizeof(int16_t))/8) |
| |
| |
| class AudioHardware : public AudioHardwareBase |
| { |
| class AudioStreamOutALSA; |
| class AudioStreamInALSA; |
| public: |
| |
| // input path names used to translate from input sources to driver paths |
| static const char *inputPathNameDefault; |
| static const char *inputPathNameCamcorder; |
| static const char *inputPathNameVoiceRecognition; |
| static const char *inputPathNameVoiceCommunication; |
| |
| AudioHardware(); |
| virtual ~AudioHardware(); |
| virtual status_t initCheck(); |
| |
| virtual status_t setVoiceVolume(float volume); |
| virtual status_t setMasterVolume(float volume); |
| |
| virtual status_t setMode(int mode); |
| |
| virtual status_t setMicMute(bool state); |
| virtual status_t getMicMute(bool* state); |
| |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| |
| virtual AudioStreamOut* openOutputStream( |
| uint32_t devices, int *format=0, uint32_t *channels=0, |
| uint32_t *sampleRate=0, status_t *status=0); |
| |
| virtual AudioStreamIn* openInputStream( |
| uint32_t devices, int *format, uint32_t *channels, |
| uint32_t *sampleRate, status_t *status, |
| AudioSystem::audio_in_acoustics acoustics); |
| |
| virtual void closeOutputStream(AudioStreamOut* out); |
| virtual void closeInputStream(AudioStreamIn* in); |
| |
| virtual size_t getInputBufferSize( |
| uint32_t sampleRate, int format, int channelCount); |
| |
| int mode() { return mMode; } |
| const char *getOutputRouteFromDevice(uint32_t device); |
| const char *getInputRouteFromDevice(uint32_t device); |
| const char *getVoiceRouteFromDevice(uint32_t device); |
| |
| status_t setIncallPath_l(uint32_t device); |
| |
| status_t setInputSource_l(audio_source source); |
| |
| static uint32_t getInputSampleRate(uint32_t sampleRate); |
| sp <AudioStreamInALSA> getActiveInput_l(); |
| |
| Mutex& lock() { return mLock; } |
| |
| struct pcm *openPcmOut_l(); |
| void closePcmOut_l(); |
| |
| struct mixer *openMixer_l(); |
| void closeMixer_l(); |
| |
| sp <AudioStreamOutALSA> output() { return mOutput; } |
| |
| protected: |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| private: |
| |
| enum tty_modes { |
| TTY_MODE_OFF, |
| TTY_MODE_VCO, |
| TTY_MODE_HCO, |
| TTY_MODE_FULL |
| }; |
| |
| bool mInit; |
| bool mMicMute; |
| sp <AudioStreamOutALSA> mOutput; |
| SortedVector < sp<AudioStreamInALSA> > mInputs; |
| Mutex mLock; |
| struct pcm* mPcm; |
| struct mixer* mMixer; |
| uint32_t mPcmOpenCnt; |
| uint32_t mMixerOpenCnt; |
| bool mInCallAudioMode; |
| |
| audio_source mInputSource; |
| bool mBluetoothNrec; |
| int mTTYMode; |
| |
| void* mSecRilLibHandle; |
| HRilClient mRilClient; |
| bool mActivatedCP; |
| HRilClient (*openClientRILD) (void); |
| int (*disconnectRILD) (HRilClient); |
| int (*closeClientRILD) (HRilClient); |
| int (*isConnectedRILD) (HRilClient); |
| int (*connectRILD) (HRilClient); |
| int (*setCallVolume) (HRilClient, SoundType, int); |
| int (*setCallAudioPath)(HRilClient, AudioPath); |
| int (*setCallClockSync)(HRilClient, SoundClockCondition); |
| void loadRILD(void); |
| status_t connectRILDIfRequired(void); |
| |
| // trace driver operations for dump |
| int mDriverOp; |
| |
| static uint32_t checkInputSampleRate(uint32_t sampleRate); |
| static const uint32_t inputSamplingRates[]; |
| |
| class AudioStreamOutALSA : public AudioStreamOut, public RefBase |
| { |
| public: |
| AudioStreamOutALSA(); |
| virtual ~AudioStreamOutALSA(); |
| status_t set(AudioHardware* mHardware, |
| uint32_t devices, |
| int *pFormat, |
| uint32_t *pChannels, |
| uint32_t *pRate); |
| virtual uint32_t sampleRate() |
| const { return mSampleRate; } |
| virtual size_t bufferSize() |
| const { return mBufferSize; } |
| virtual uint32_t channels() |
| const { return mChannels; } |
| virtual int format() |
| const { return AUDIO_HW_OUT_FORMAT; } |
| virtual uint32_t latency() |
| const { return (1000 * AUDIO_HW_OUT_PERIOD_CNT * |
| (bufferSize()/frameSize()))/sampleRate() + |
| AUDIO_HW_OUT_LATENCY_MS; } |
| virtual status_t setVolume(float left, float right) |
| { return INVALID_OPERATION; } |
| virtual ssize_t write(const void* buffer, size_t bytes); |
| virtual status_t standby(); |
| bool checkStandby(); |
| |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| uint32_t device() { return mDevices; } |
| virtual status_t getRenderPosition(uint32_t *dspFrames); |
| |
| void doStandby_l(); |
| void close_l(); |
| status_t open_l(); |
| int standbyCnt() { return mStandbyCnt; } |
| |
| int prepareLock(); |
| void lock(); |
| void unlock(); |
| |
| private: |
| |
| Mutex mLock; |
| AudioHardware* mHardware; |
| struct pcm *mPcm; |
| struct mixer *mMixer; |
| struct mixer_ctl *mRouteCtl; |
| const char *next_route; |
| bool mStandby; |
| uint32_t mDevices; |
| uint32_t mChannels; |
| uint32_t mSampleRate; |
| size_t mBufferSize; |
| // trace driver operations for dump |
| int mDriverOp; |
| int mStandbyCnt; |
| bool mSleepReq; |
| }; |
| |
| class DownSampler; |
| |
| class BufferProvider |
| { |
| public: |
| |
| struct Buffer { |
| union { |
| void* raw; |
| short* i16; |
| int8_t* i8; |
| }; |
| size_t frameCount; |
| }; |
| |
| virtual ~BufferProvider() {} |
| |
| virtual status_t getNextBuffer(Buffer* buffer) = 0; |
| virtual void releaseBuffer(Buffer* buffer) = 0; |
| }; |
| |
| class DownSampler { |
| public: |
| DownSampler(uint32_t outSampleRate, |
| uint32_t channelCount, |
| uint32_t frameCount, |
| BufferProvider* provider); |
| |
| virtual ~DownSampler(); |
| |
| void reset(); |
| status_t initCheck() { return mStatus; } |
| int resample(int16_t* out, size_t *outFrameCount); |
| |
| private: |
| status_t mStatus; |
| BufferProvider* mProvider; |
| uint32_t mSampleRate; |
| uint32_t mChannelCount; |
| uint32_t mFrameCount; |
| int16_t *mInLeft; |
| int16_t *mInRight; |
| int16_t *mTmpLeft; |
| int16_t *mTmpRight; |
| int16_t *mTmp2Left; |
| int16_t *mTmp2Right; |
| int16_t *mOutLeft; |
| int16_t *mOutRight; |
| int mInInBuf; |
| int mInTmpBuf; |
| int mInTmp2Buf; |
| int mOutBufPos; |
| int mInOutBuf; |
| }; |
| |
| |
| class AudioStreamInALSA : public AudioStreamIn, public BufferProvider, public RefBase |
| { |
| |
| public: |
| AudioStreamInALSA(); |
| virtual ~AudioStreamInALSA(); |
| status_t set(AudioHardware* hw, |
| uint32_t devices, |
| int *pFormat, |
| uint32_t *pChannels, |
| uint32_t *pRate, |
| AudioSystem::audio_in_acoustics acoustics); |
| virtual size_t bufferSize() const { return mBufferSize; } |
| virtual uint32_t channels() const { return mChannels; } |
| virtual int format() const { return AUDIO_HW_IN_FORMAT; } |
| virtual uint32_t sampleRate() const { return mSampleRate; } |
| virtual status_t setGain(float gain) { return INVALID_OPERATION; } |
| virtual ssize_t read(void* buffer, ssize_t bytes); |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| virtual status_t standby(); |
| bool checkStandby(); |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| virtual unsigned int getInputFramesLost() const { return 0; } |
| uint32_t device() { return mDevices; } |
| void doStandby_l(); |
| void close_l(); |
| status_t open_l(); |
| int standbyCnt() { return mStandbyCnt; } |
| |
| static size_t getBufferSize(uint32_t sampleRate, int channelCount); |
| |
| // BufferProvider |
| virtual status_t getNextBuffer(BufferProvider::Buffer* buffer); |
| virtual void releaseBuffer(BufferProvider::Buffer* buffer); |
| |
| int prepareLock(); |
| void lock(); |
| void unlock(); |
| |
| private: |
| Mutex mLock; |
| AudioHardware* mHardware; |
| struct pcm *mPcm; |
| struct mixer *mMixer; |
| struct mixer_ctl *mRouteCtl; |
| const char *next_route; |
| bool mStandby; |
| uint32_t mDevices; |
| uint32_t mChannels; |
| uint32_t mChannelCount; |
| uint32_t mSampleRate; |
| size_t mBufferSize; |
| DownSampler *mDownSampler; |
| status_t mReadStatus; |
| size_t mInPcmInBuf; |
| int16_t *mPcmIn; |
| // trace driver operations for dump |
| int mDriverOp; |
| int mStandbyCnt; |
| bool mSleepReq; |
| }; |
| |
| }; |
| |
| }; // namespace android |
| |
| #endif |