| /* |
| ** 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> |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| // Kernel driver interface |
| // |
| /* Source (TX) devices */ |
| #define ADSP_AUDIO_DEVICE_ID_HANDSET_MIC 0x107ac8d |
| #define ADSP_AUDIO_DEVICE_ID_HEADSET_MIC 0x1081510 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MIC 0x1081512 |
| #define ADSP_AUDIO_DEVICE_ID_BT_SCO_MIC 0x1081518 |
| #define ADSP_AUDIO_DEVICE_ID_TTY_HEADSET_MIC 0x108151b |
| #define ADSP_AUDIO_DEVICE_ID_I2S_MIC 0x1089bf3 |
| |
| /* Special loopback pseudo device to be paired with an RX device */ |
| /* with usage ADSP_AUDIO_DEVICE_USAGE_MIXED_PCM_LOOPBACK */ |
| #define ADSP_AUDIO_DEVICE_ID_MIXED_PCM_LOOPBACK_TX 0x1089bf2 |
| |
| /* Sink (RX) devices */ |
| #define ADSP_AUDIO_DEVICE_ID_HANDSET_SPKR 0x107ac88 |
| #define ADSP_AUDIO_DEVICE_ID_HEADSET_SPKR_MONO 0x1081511 |
| #define ADSP_AUDIO_DEVICE_ID_HEADSET_SPKR_STEREO 0x107ac8a |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO 0x1081513 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO_W_MONO_HEADSET 0x108c508 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO_W_STEREO_HEADSET 0x108c894 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_STEREO 0x1081514 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_STEREO_W_MONO_HEADSET 0x108c895 |
| #define ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_STEREO_W_STEREO_HEADSET 0x108c509 |
| #define ADSP_AUDIO_DEVICE_ID_BT_SCO_SPKR 0x1081519 |
| #define ADSP_AUDIO_DEVICE_ID_TTY_HEADSET_SPKR 0x108151c |
| #define ADSP_AUDIO_DEVICE_ID_I2S_SPKR 0x1089bf4 |
| |
| #define HANDSET_MIC ADSP_AUDIO_DEVICE_ID_HANDSET_MIC |
| #define HANDSET_SPKR ADSP_AUDIO_DEVICE_ID_HANDSET_SPKR |
| #define HEADSET_MIC ADSP_AUDIO_DEVICE_ID_HEADSET_MIC |
| #define HEADSET_SPKR_MONO ADSP_AUDIO_DEVICE_ID_HEADSET_SPKR_MONO |
| #define HEADSET_SPKR_STEREO ADSP_AUDIO_DEVICE_ID_HEADSET_SPKR_STEREO |
| #define SPKR_PHONE_MIC ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MIC |
| #define SPKR_PHONE_MONO ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO |
| #define SPKR_PHONE_STEREO ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_STEREO |
| #define BT_A2DP_SPKR ADSP_AUDIO_DEVICE_ID_BT_A2DP_SPKR |
| #define BT_SCO_MIC ADSP_AUDIO_DEVICE_ID_BT_SCO_MIC |
| #define BT_SCO_SPKR ADSP_AUDIO_DEVICE_ID_BT_SCO_SPKR |
| #define TTY_HEADSET_MIC ADSP_AUDIO_DEVICE_ID_TTY_HEADSET_MIC |
| #define TTY_HEADSET_SPKR ADSP_AUDIO_DEVICE_ID_TTY_HEADSET_SPKR |
| #define FM_HEADSET ADSP_AUDIO_DEVICE_ID_HEADSET_SPKR_STEREO |
| #define FM_SPKR ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO |
| #define SPKR_PHONE_HEADSET_STEREO ADSP_AUDIO_DEVICE_ID_SPKR_PHONE_MONO_W_MONO_HEADSET |
| |
| #define ACDB_ID_HAC_HANDSET_MIC 107 |
| #define ACDB_ID_HAC_HANDSET_SPKR 207 |
| #define ACDB_ID_EXT_MIC_REC 307 |
| #define ACDB_ID_HEADSET_PLAYBACK 407 |
| #define ACDB_ID_HEADSET_RINGTONE_PLAYBACK 408 |
| #define ACDB_ID_INT_MIC_REC 507 |
| #define ACDB_ID_CAMCORDER 508 |
| #define ACDB_ID_INT_MIC_VR 509 |
| #define ACDB_ID_SPKR_PLAYBACK 607 |
| #define ACDB_ID_ALT_SPKR_PLAYBACK 609 |
| |
| #define SAMP_RATE_INDX_8000 0 |
| #define SAMP_RATE_INDX_11025 1 |
| #define SAMP_RATE_INDX_12000 2 |
| #define SAMP_RATE_INDX_16000 3 |
| #define SAMP_RATE_INDX_22050 4 |
| #define SAMP_RATE_INDX_24000 5 |
| #define SAMP_RATE_INDX_32000 6 |
| #define SAMP_RATE_INDX_44100 7 |
| #define SAMP_RATE_INDX_48000 8 |
| |
| #define EQ_MAX_BAND_NUM 12 |
| |
| #define ADRC_ENABLE 0x0001 |
| #define ADRC_DISABLE 0x0000 |
| #define EQ_ENABLE 0x0002 |
| #define EQ_DISABLE 0x0000 |
| #define RX_IIR_ENABLE 0x0004 |
| #define RX_IIR_DISABLE 0x0000 |
| |
| #define MOD_PLAY 1 |
| #define MOD_REC 2 |
| |
| struct msm_bt_endpoint { |
| int tx; |
| int rx; |
| char name[64]; |
| }; |
| |
| struct eq_filter_type { |
| int16_t gain; |
| uint16_t freq; |
| uint16_t type; |
| uint16_t qf; |
| }; |
| |
| struct eqalizer { |
| uint16_t bands; |
| uint16_t params[132]; |
| }; |
| |
| struct rx_iir_filter { |
| uint16_t num_bands; |
| uint16_t iir_params[48]; |
| }; |
| |
| struct msm_audio_config { |
| uint32_t buffer_size; |
| uint32_t buffer_count; |
| uint32_t channel_count; |
| uint32_t sample_rate; |
| uint32_t codec_type; |
| uint32_t unused[3]; |
| }; |
| |
| struct msm_mute_info { |
| uint32_t mute; |
| uint32_t path; |
| }; |
| |
| #define CODEC_TYPE_PCM 0 |
| #define PCM_FILL_BUFFER_COUNT 1 |
| #define AUDIO_HW_NUM_OUT_BUF 4 // Number of buffers in audio driver for output |
| // TODO: determine actual audio DSP and hardware latency |
| #define AUDIO_HW_OUT_LATENCY_MS 0 // Additionnal latency introduced by audio DSP and hardware in ms |
| #define AUDIO_HW_OUT_SAMPLERATE 44100 // Default audio output sample rate |
| #define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO) // Default audio output channel mask |
| #define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT) // Default audio output sample format |
| #define AUDIO_HW_OUT_BUFSZ 3072 // Default audio output buffer size |
| |
| #define AUDIO_HW_IN_SAMPLERATE 8000 // Default audio input sample rate |
| #define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_MONO) // Default audio input channel mask |
| #define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT) // Default audio input sample format |
| #define AUDIO_HW_IN_BUFSZ 256 // Default audio input buffer size |
| |
| #define VOICE_VOLUME_MAX 5 // Maximum voice volume |
| // ---------------------------------------------------------------------------- |
| |
| |
| class AudioHardware : public AudioHardwareBase |
| { |
| class AudioStreamOutMSM72xx; |
| class AudioStreamInMSM72xx; |
| |
| public: |
| 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); |
| |
| // mic mute |
| 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); |
| |
| // create I/O streams |
| 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); |
| |
| void clearCurDevice() { mCurSndDevice = -1; } |
| |
| protected: |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| private: |
| |
| status_t doAudioRouteOrMute(uint32_t device); |
| status_t setMicMute_nosync(bool state); |
| status_t checkMicMute(); |
| status_t dumpInternals(int fd, const Vector<String16>& args); |
| uint32_t getInputSampleRate(uint32_t sampleRate); |
| bool checkOutputStandby(); |
| status_t get_mMode(); |
| status_t get_mRoutes(); |
| status_t set_mRecordState(bool onoff); |
| status_t doA1026_init(); |
| status_t get_snd_dev(); |
| status_t get_batt_temp(int *batt_temp); |
| status_t doAudience_A1026_Control(int Mode, bool Record, uint32_t Routes); |
| status_t doRouting(); |
| status_t updateACDB(); |
| uint32_t getACDB(int mode, int device); |
| AudioStreamInMSM72xx* getActiveInput_l(); |
| status_t do_tpa2018_control(int mode); |
| size_t getBufferSize(uint32_t sampleRate, int channelCount); |
| |
| class AudioStreamOutMSM72xx : public AudioStreamOut { |
| public: |
| AudioStreamOutMSM72xx(); |
| virtual ~AudioStreamOutMSM72xx(); |
| status_t set(AudioHardware* mHardware, |
| uint32_t devices, |
| int *pFormat, |
| uint32_t *pChannels, |
| uint32_t *pRate); |
| virtual uint32_t sampleRate() const { return mSampleRate; } |
| // must be 32-bit aligned |
| 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_NUM_OUT_BUF*(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(); |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| bool checkStandby(); |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| uint32_t devices() { return mDevices; } |
| virtual status_t getRenderPosition(uint32_t *dspFrames); |
| |
| private: |
| AudioHardware* mHardware; |
| int mFd; |
| int mStartCount; |
| int mRetryCount; |
| bool mStandby; |
| uint32_t mDevices; |
| uint32_t mChannels; |
| uint32_t mSampleRate; |
| size_t mBufferSize; |
| }; |
| |
| class AudioStreamInMSM72xx : public AudioStreamIn { |
| public: |
| AudioStreamInMSM72xx(); |
| virtual ~AudioStreamInMSM72xx(); |
| status_t set(AudioHardware* mHardware, |
| 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 mFormat; } |
| 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(); |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| virtual unsigned int getInputFramesLost() const { return 0; } |
| uint32_t devices() { return mDevices; } |
| bool checkStandby(); |
| |
| private: |
| AudioHardware* mHardware; |
| int mFd; |
| bool mStandby; |
| int mRetryCount; |
| int mFormat; |
| uint32_t mChannels; |
| uint32_t mSampleRate; |
| size_t mBufferSize; |
| AudioSystem::audio_in_acoustics mAcoustics; |
| uint32_t mDevices; |
| }; |
| |
| enum tty_modes { |
| TTY_MODE_OFF, |
| TTY_MODE_FULL, |
| TTY_MODE_VCO, |
| TTY_MODE_HCO |
| }; |
| |
| static const uint32_t inputSamplingRates[]; |
| Mutex mA1026Lock; |
| bool mA1026Init; |
| bool mRecordState; |
| bool mInit; |
| bool mMicMute; |
| bool mBluetoothNrec; |
| bool mHACSetting; |
| uint32_t mBluetoothIdTx; |
| uint32_t mBluetoothIdRx; |
| AudioStreamOutMSM72xx* mOutput; |
| SortedVector <AudioStreamInMSM72xx*> mInputs; |
| |
| msm_bt_endpoint *mBTEndpoints; |
| int mNumBTEndpoints; |
| int mCurSndDevice; |
| int mNoiseSuppressionState; |
| uint32_t mVoiceVolume; |
| |
| friend class AudioStreamInMSM72xx; |
| Mutex mLock; |
| uint32_t mRoutes[AudioSystem::NUM_MODES]; |
| int mTTYMode; |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_HARDWARE_MSM72XX_H |
