/*
** 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