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/* AudioHardwareALSA.h
**
** Copyright 2008, Wind River Systems
**
** 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_ALSA_H
#define ANDROID_AUDIO_HARDWARE_ALSA_H
#include <stdint.h>
#include <sys/types.h>
#include <alsa/asoundlib.h>
#include <hardware_legacy/AudioHardwareBase.h>
#if defined SEC_IPC
#include <hardware/hardware.h>
// sangsu fix : headers for IPC
#include "secril-client.h"
// sangsu fix : defines for IPC
#define OEM_FUNCTION_ID_SOUND 0x08 // sound Main Cmd
//sangsu fix : sound sub command for IPC
#define OEM_SOUND_SET_VOLUME_CTRL 0x03
#define OEM_SOUND_GET_VOLUME_CTRL 0x04
#define OEM_SOUND_SET_AUDIO_PATH_CTRL 0x05
#define OEM_SOUND_GET_AUDIO_PATH_CTRL 0x06
//sangsu fix : audio path for IPC
#define OEM_SOUND_AUDIO_PATH_HANDSET 0x01
#define OEM_SOUND_AUDIO_PATH_HEADSET 0x02
#define OEM_SOUND_AUDIO_PATH_HANDFREE 0x03
#define OEM_SOUND_AUDIO_PATH_BLUETOOTH 0x04
#define OEM_SOUND_AUDIO_PATH_STREOBT 0x05
#define OEM_SOUND_AUDIO_PATH_SPEAKER 0x06
#define OEM_SOUND_AUDIO_PATH_HEADSET35 0x07
#define OEM_SOUND_AUDIO_PATH_BT_NSEC_OFF 0x08
// sangsu fix : volume level for IPC
#define OEM_SOUND_VOLUME_LEVEL_MUTE 0x00
#define OEM_SOUND_VOLUME_LEVEL1 0x01
#define OEM_SOUND_VOLUME_LEVEL2 0x02
#define OEM_SOUND_VOLUME_LEVEL3 0x03
#define OEM_SOUND_VOLUME_LEVEL4 0x04
#define OEM_SOUND_VOLUME_LEVEL5 0x05
#define OEM_SOUND_VOLUME_LEVEL6 0x06
#define OEM_SOUND_VOLUME_LEVEL7 0x07
#define OEM_SOUND_VOLUME_LEVEL8 0x08
// For synchronizing I2S clocking
#if defined SYNCHRONIZE_CP
#define OEM_SOUND_SET_CLOCK_CTRL 0x0A
#define OEM_SOUND_CLOCK_START 0x01
#define OEM_SOUND_CLOCK_STOP 0x00
#endif
// For VT
#if defined VIDEO_TELEPHONY
#define OEM_SOUND_VIDEO_CALL_STOP 0x00
#define OEM_SOUND_VIDEO_CALL_START 0x01
#define OEM_SOUND_SET_VIDEO_CALL_CTRL 0x07
#endif
// sangsu fix : volume type for IPC
#define OEM_SOUND_TYPE_VOICE 0x01 // Receiver(0x00) + Voice(0x01)
#define OEM_SOUND_TYPE_KEYTONE 0x02 // Receiver(0x00) + Key tone (0x02)
#define OEM_SOUND_TYPE_BELL 0x03 // Receiver(0x00) + Bell (0x03)
#define OEM_SOUND_TYPE_MESSAGE 0x04 // Receiver(0x00) + Message(0x04)
#define OEM_SOUND_TYPE_ALARM 0x05 // Receiver(0x00) + Alarm (0x05)
#define OEM_SOUND_TYPE_SPEAKER 0x11 // SpeakerPhone (0x10) + Voice(0x01)
#define OEM_SOUND_TYPE_HFKVOICE 0x21 // HFK (0x20) + Voice(0x01)
#define OEM_SOUND_TYPE_HFKKEY 0x22 // HFK (0x20) + Key tone (0x02)
#define OEM_SOUND_TYPE_HFKBELL 0x23 // HFK (0x20) + Bell (0x03)
#define OEM_SOUND_TYPE_HFKMSG 0x24 // HFK (0x20) + Message(0x04)
#define OEM_SOUND_TYPE_HFKALARM 0x25 // HFK (0x20) + Alarm (0x05)
#define OEM_SOUND_TYPE_HFKPDA 0x26 // HFK (0x20) + PDA miscellaneous sound (0x06)
#define OEM_SOUND_TYPE_HEADSET 0x31 // Headset (0x30) + Voice(0x01)
#define OEM_SOUND_TYPE_BTVOICE 0x41 // BT(0x40) + Voice(0x01)
#endif
#ifndef ALSA_DEFAULT_SAMPLE_RATE
#define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
#endif
#define DEFAULT_SAMPLE_RATE ALSA_DEFAULT_SAMPLE_RATE
#define PLAYBACK 0
#define PERIOD_SZ_PLAYBACK 1024
#define PERIODS_PLAYBACK 4
#define BUFFER_SZ_PLAYBACK (PERIODS_PLAYBACK * PERIOD_SZ_PLAYBACK)
#define LATENCY_PLAYBACK_MS ((BUFFER_SZ_PLAYBACK * 1000 / DEFAULT_SAMPLE_RATE) * 1000)
#define CAPTURE 1
#define PERIOD_SZ_CAPTURE 2048
#define PERIODS_CAPTURE 2
#define BUFFER_SZ_CAPTURE (PERIODS_CAPTURE * PERIOD_SZ_CAPTURE)
#define LATENCY_CAPTURE_MS ((BUFFER_SZ_CAPTURE * 1000 / DEFAULT_SAMPLE_RATE) * 1000)
namespace android
{
class AudioHardwareALSA;
// ----------------------------------------------------------------------------
class ALSAMixer
{
public:
ALSAMixer();
virtual ~ALSAMixer();
bool isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
status_t setMasterVolume(float volume);
status_t setMasterGain(float gain);
status_t setVolume(uint32_t device, float volume);
status_t setGain(uint32_t device, float gain);
status_t setCaptureMuteState(uint32_t device, bool state);
status_t getCaptureMuteState(uint32_t device, bool *state);
status_t setPlaybackMuteState(uint32_t device, bool state);
status_t getPlaybackMuteState(uint32_t device, bool *state);
private:
snd_mixer_t *mMixer[SND_PCM_STREAM_LAST+1];
};
class ALSAControl
{
public:
ALSAControl(const char *device = "default");
virtual ~ALSAControl();
status_t get(const char *name, unsigned int &value, int index = 0);
status_t set(const char *name, unsigned int value, int index = -1);
private:
snd_ctl_t *mHandle;
};
class ALSAStreamOps
{
protected:
friend class AudioStreamOutALSA;
friend class AudioStreamInALSA;
struct StreamDefaults
{
const char * devicePrefix;
snd_pcm_stream_t direction; // playback or capture
snd_pcm_format_t format;
int channelCount;
uint32_t sampleRate;
uint32_t smpRateShift;
unsigned int latency; // Delay in usec
unsigned int bufferSize; // Size of sample buffer
unsigned int periodSize; // Size of sample buffer
};
ALSAStreamOps();
virtual ~ALSAStreamOps();
status_t set(int *format,
uint32_t *channels,
uint32_t *rate);
virtual uint32_t sampleRate() const;
status_t sampleRate(uint32_t rate);
virtual size_t bufferSize() const;
virtual int format() const;
int getAndroidFormat(snd_pcm_format_t format);
virtual uint32_t channels() const;
int channelCount() const;
status_t channelCount(int channelCount);
uint32_t getAndroidChannels(int channelCount) const;
status_t open(int mode, uint32_t device);
void close();
status_t setSoftwareParams();
status_t setPCMFormat(snd_pcm_format_t format);
status_t setHardwareResample(bool resample);
const char *streamName();
virtual status_t setDevice(int mode, uint32_t device, uint32_t audio_mode);
const char *deviceName(int mode, uint32_t device);
void setStreamDefaults(StreamDefaults *dev) {
mDefaults = dev;
}
Mutex mLock;
private:
snd_pcm_t *mHandle;
snd_pcm_hw_params_t *mHardwareParams;
snd_pcm_sw_params_t *mSoftwareParams;
int mMode;
uint32_t mDevice;
StreamDefaults *mDefaults;
};
// ----------------------------------------------------------------------------
class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
{
public:
AudioStreamOutALSA(AudioHardwareALSA *parent);
virtual ~AudioStreamOutALSA();
status_t set(int *format,
uint32_t *channelCount,
uint32_t *sampleRate){
return ALSAStreamOps::set(format, channelCount, sampleRate);
}
virtual uint32_t sampleRate() const {
return ALSAStreamOps::sampleRate();
}
virtual size_t bufferSize() const
{
return ALSAStreamOps::bufferSize();
}
virtual uint32_t channels() const
{
return ALSAStreamOps::channels();
}
virtual int format() const
{
return ALSAStreamOps::format();
}
virtual uint32_t latency() const;
virtual ssize_t write(const void *buffer, size_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
virtual status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode);
virtual status_t setVolume(float left, float right); //Tushar: New arch
status_t setVolume(float volume);
status_t standby();
bool isStandby();
void setWakeLock();
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys);
virtual status_t getRenderPosition(uint32_t *dspFrames);
private:
AudioHardwareALSA *mParent;
bool mPowerLock;
};
class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
{
public:
AudioStreamInALSA(AudioHardwareALSA *parent);
virtual ~AudioStreamInALSA();
status_t set(int *format,
uint32_t *channelCount,
uint32_t *sampleRate){
return ALSAStreamOps::set(format, channelCount, sampleRate);
}
virtual uint32_t sampleRate() const {
return ALSAStreamOps::sampleRate();
}
virtual size_t bufferSize() const
{
return ALSAStreamOps::bufferSize();
}
virtual uint32_t channels() const
{
return ALSAStreamOps::channels();
}
virtual int format() const
{
return ALSAStreamOps::format();
}
virtual ssize_t read(void* buffer, ssize_t bytes);
virtual status_t dump(int fd, const Vector<String16>& args);
virtual status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode);
virtual status_t setGain(float gain);
virtual status_t standby();
status_t standby_l();
void setWakeLock();
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys);
virtual unsigned int getInputFramesLost() const { return 0; }
private:
AudioHardwareALSA *mParent;
bool mPowerLock;
int16_t mBuffer[2 * PERIOD_SZ_CAPTURE];
};
#if defined SEC_IPC
//TODO..implementation has to be done
class AudioHardwareIPC
{
public:
AudioHardwareIPC();
virtual ~AudioHardwareIPC();
status_t transmitVolumeIPC(uint32_t type, float volume);
status_t transmitAudioPathIPC(uint32_t path);
#if defined SYNCHRONIZE_CP
status_t transmitClock_IPC(uint32_t condition);
#endif
private:
HRilClient mClient;
char data[100];
};
#endif
class AudioHardwareALSA : public AudioHardwareBase
{
public:
AudioHardwareALSA();
virtual ~AudioHardwareALSA();
/**
* check to see if the audio hardware interface has been initialized.
* return status based on values defined in include/utils/Errors.h
*/
virtual status_t initCheck();
/**
* put the audio hardware into standby mode to conserve power. Returns
* status based on include/utils/Errors.h
*/
virtual status_t standby();
/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
virtual status_t setVoiceVolume(float volume);
/**
* set the audio volume for all audio activities other than voice call.
* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
* the software mixer will emulate this capability.
*/
virtual status_t setMasterVolume(float volume);
// mic mute
virtual status_t setMicMute(bool state);
virtual status_t getMicMute(bool* state);
virtual size_t getInputBufferSize(
uint32_t sampleRate,
int format,
int channelCount);
#if defined TURN_ON_DEVICE_ONLY_USE
virtual int setMicStatus(int on); // To deliver status of input stream(activated or not). If it's activated, doesn't turn off codec.
#endif
/** This method creates and opens the audio hardware output stream */
virtual AudioStreamOut* openOutputStream(
uint32_t devices,
int *format=0,
uint32_t *channels=0,
uint32_t *sampleRate=0,
status_t *status=0);
virtual void closeOutputStream(AudioStreamOut* out);
/** This method creates and opens the audio hardware input stream */
virtual AudioStreamIn* openInputStream(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status,
AudioSystem::audio_in_acoustics acoustics);
virtual void closeInputStream(AudioStreamIn* in);
static uint32_t checkInputSampleRate(uint32_t sampleRate);
static const uint32_t inputSamplingRates[];
protected:
/**
* doRouting actually initiates the routing. A call to setRouting
* or setMode may result in a routing change. The generic logic calls
* doRouting when required. If the device has any special requirements these
* methods can be overriden.
*/
virtual status_t doRouting(uint32_t device);
virtual status_t dump(int fd, const Vector<String16>& args);
friend class AudioStreamOutALSA;
friend class AudioStreamInALSA;
ALSAMixer *mMixer;
AudioStreamOutALSA *mOutput;
AudioStreamInALSA *mInput;
#if defined SEC_IPC
AudioHardwareIPC *mIPC; //for IPC
uint32_t mRoutes[AudioSystem::NUM_MODES];
#endif
private:
Mutex mLock;
#if defined TURN_ON_DEVICE_ONLY_USE
bool mActivatedInputDevice;
#endif
};
// ----------------------------------------------------------------------------
#if defined SEC_IPC
// sangsu fix : global functions for IPC
static int onRawReqComplete(HRilClient client, const void *data, size_t datalen);
static int onUnsol(HRilClient client, const void *data, size_t datalen);
#endif
}; // namespace android
#endif // ANDROID_AUDIO_HARDWARE_ALSA_H
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