diff options
Diffstat (limited to 'libaudio2/AudioHardware.h')
| -rw-r--r-- | libaudio2/AudioHardware.h | 214 |
1 files changed, 192 insertions, 22 deletions
diff --git a/libaudio2/AudioHardware.h b/libaudio2/AudioHardware.h index 9bdfe2f..ab81d9c 100644 --- a/libaudio2/AudioHardware.h +++ b/libaudio2/AudioHardware.h @@ -25,6 +25,8 @@ #include <hardware_legacy/AudioHardwareBase.h> +#include "secril-client.h" + extern "C" { struct pcm; struct mixer; @@ -33,11 +35,6 @@ extern "C" { namespace android { -#define CODEC_TYPE_PCM 0 -#define PCM_FILL_BUFFER_COUNT 1 -// Number of buffers in audio driver for output -#define AUDIO_HW_NUM_OUT_BUF 2 - // 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 @@ -47,27 +44,36 @@ namespace android { #define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO) // Default audio output sample format #define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT) -// Default audio output buffer size -#define AUDIO_HW_OUT_BUFSZ 4096 +// 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)) -#if 0 // 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) -// Default audio input buffer size -#define AUDIO_HW_IN_BUFSZ 256 +// 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) -// Maximum voice volume -#define VOICE_VOLUME_MAX 5 -#endif +#define VOICE_REC_MODE_KEY "vr_mode" class AudioHardware : public AudioHardwareBase { class AudioStreamOutALSA; + class AudioStreamInALSA; public: + AudioHardware(); virtual ~AudioHardware(); virtual status_t initCheck(); @@ -98,17 +104,66 @@ public: virtual size_t getInputBufferSize( uint32_t sampleRate, int format, int channelCount); - void clearCurDevice() { } + 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(uint32_t device); + status_t setIncallPath_l(uint32_t device); + + status_t setVoiceRecognition(bool enable); + status_t setVoiceRecognition_l(bool enable); + + static uint32_t getInputSampleRate(uint32_t sampleRate); + AudioStreamInALSA* getActiveInput_l(); + + void lock() { mLock.lock(); } + void unlock() { mLock.unlock(); } + + struct pcm *openPcmOut(); + struct pcm *openPcmOut_l(); + void closePcmOut(); + void closePcmOut_l(); + + struct mixer *openMixer(); + struct mixer *openMixer_l(); + void closeMixer(); + void closeMixer_l(); protected: virtual status_t dump(int fd, const Vector<String16>& args); private: - bool mInit; - bool mMicMute; - AudioStreamOutALSA *mOutput; - Mutex mLock; - struct mixer *mMixer; + + bool mInit; + bool mMicMute; + AudioStreamOutALSA* mOutput; + SortedVector <AudioStreamInALSA*> mInputs; + Mutex mLock; + struct pcm* mPcm; + struct mixer* mMixer; + uint32_t mPcmOpenCnt; + uint32_t mMixerOpenCnt; + + bool mVrModeEnabled; + bool mBluetoothNrec; + 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); + + static uint32_t checkInputSampleRate(uint32_t sampleRate); + static const uint32_t inputSamplingRates[]; class AudioStreamOutALSA : public AudioStreamOut { @@ -129,7 +184,7 @@ private: virtual int format() const { return AUDIO_HW_OUT_FORMAT; } virtual uint32_t latency() - const { return (1000 * AUDIO_HW_NUM_OUT_BUF * + const { return (1000 * AUDIO_HW_OUT_PERIOD_CNT * (bufferSize()/frameSize()))/sampleRate() + AUDIO_HW_OUT_LATENCY_MS; } virtual status_t setVolume(float left, float right) @@ -140,11 +195,14 @@ private: bool checkStandby(); virtual status_t setParameters(const String8& keyValuePairs); virtual String8 getParameters(const String8& keys); - uint32_t devices() - { return mDevices; } + uint32_t device() { return mDevices; } virtual status_t getRenderPosition(uint32_t *dspFrames); + void setNextRoute(const char *route) { next_route = route; } + private: + + Mutex mLock; AudioHardware* mHardware; struct pcm *mPcm; struct mixer *mMixer; @@ -158,6 +216,118 @@ private: uint32_t mSampleRate; size_t mBufferSize; }; + + 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: + 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 setNextRoute(const char *route) { next_route = route; } + + static size_t getBufferSize(uint32_t sampleRate, int channelCount); + + // BufferProvider + virtual status_t getNextBuffer(BufferProvider::Buffer* buffer); + virtual void releaseBuffer(BufferProvider::Buffer* buffer); + + private: + Mutex mLock; + AudioHardware* mHardware; + struct pcm *mPcm; + struct mixer *mMixer; + struct mixer_ctl *mRouteCtl; + struct mixer_ctl *mMicCtl; + const char *next_route; + int mStartCount; + int mRetryCount; + 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; + }; + }; }; // namespace android |