/* ** 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> extern "C" { #include <linux/msm_audio.h> } namespace android { // ---------------------------------------------------------------------------- // Kernel driver interface // #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 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_audio_stats { uint32_t out_bytes; uint32_t unused[3]; }; #define CODEC_TYPE_PCM 0 #define AUDIO_HW_NUM_OUT_BUF 2 // 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_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_BUFFERSIZE 2048 // Default audio input buffer size #define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT) // Default audio input sample format // ---------------------------------------------------------------------------- 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 doRouting(); AudioStreamInMSM72xx* getActiveInput_l(); 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 44100; } // must be 32-bit aligned - driver only seems to like 4800 virtual size_t bufferSize() const { return 4800; } virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; } virtual int format() const { return AudioSystem::PCM_16_BIT; } 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; }; class AudioStreamInMSM72xx : public AudioStreamIn { public: enum input_state { AUDIO_INPUT_CLOSED, AUDIO_INPUT_OPENED, AUDIO_INPUT_STARTED }; 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; } int state() const { return mState; } private: AudioHardware* mHardware; int mFd; int mState; int mRetryCount; int mFormat; uint32_t mChannels; uint32_t mSampleRate; size_t mBufferSize; AudioSystem::audio_in_acoustics mAcoustics; uint32_t mDevices; }; static const uint32_t inputSamplingRates[]; bool mInit; bool mMicMute; bool mBluetoothNrec; uint32_t mBluetoothId; AudioStreamOutMSM72xx* mOutput; SortedVector <AudioStreamInMSM72xx*> mInputs; msm_snd_endpoint *mSndEndpoints; int mNumSndEndpoints; int mCurSndDevice; friend class AudioStreamInMSM72xx; Mutex mLock; int SND_DEVICE_CURRENT; int SND_DEVICE_HANDSET; int SND_DEVICE_SPEAKER; int SND_DEVICE_HEADSET; int SND_DEVICE_BT; int SND_DEVICE_CARKIT; int SND_DEVICE_TTY_FULL; int SND_DEVICE_TTY_VCO; int SND_DEVICE_TTY_HCO; int SND_DEVICE_NO_MIC_HEADSET; int SND_DEVICE_FM_HEADSET; int SND_DEVICE_HEADSET_AND_SPEAKER; int SND_DEVICE_FM_SPEAKER; int SND_DEVICE_BT_EC_OFF; }; // ---------------------------------------------------------------------------- }; // namespace android #endif // ANDROID_AUDIO_HARDWARE_MSM72XX_H