/*
** 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