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diff --git a/libaudio/AudioHardwareALSA.h b/libaudio/AudioHardwareALSA.h
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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>
-
-// sangsu fix : headers for IPC
-#include "secril-client.h"
-
-#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 1024
-#define PERIODS_CAPTURE 4
-#define BUFFER_SZ_CAPTURE (PERIODS_CAPTURE * PERIOD_SZ_CAPTURE)
-#define LATENCY_CAPTURE_MS ((BUFFER_SZ_CAPTURE * 1000 / DEFAULT_SAMPLE_RATE) * 1000)
-
-//Recognition param
-#define RECOGNITION_OFF 0
-#define RECOGNITION_ON 1
-
-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
- {
- public:
- uint32_t device() { return mDevice; }
- void close();
-
- 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 bufferRatio;
- 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);
- status_t setSoftwareParams();
- status_t setPCMFormat(snd_pcm_format_t format);
- status_t setHardwareResample(bool resample);
-
- const char *streamName();
- 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;
- 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);
- status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
- bool force = false);
- virtual status_t setVolume(float left, float right); //Tushar: New arch
-
- status_t setVolume(float volume);
-
- status_t standby();
-
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
-
- virtual status_t getRenderPosition(uint32_t *dspFrames);
- bool isActive() { return mPowerLock; }
-
- private:
- AudioHardwareALSA *mParent;
- bool mPowerLock;
- };
-
- class ALSADownsampler;
-
- class ALSABufferProvider
- {
- public:
-
- struct Buffer {
- union {
- void* raw;
- short* i16;
- int8_t* i8;
- };
- size_t frameCount;
- };
-
- virtual ~ALSABufferProvider() {}
-
- virtual status_t getNextBuffer(Buffer* buffer) = 0;
- virtual void releaseBuffer(Buffer* buffer) = 0;
- };
-
- class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps, public ALSABufferProvider
- {
- public:
- AudioStreamInALSA(AudioHardwareALSA *parent);
- virtual ~AudioStreamInALSA();
-
- status_t set(int *format,
- uint32_t *channelCount,
- uint32_t *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);
- status_t setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
- bool force = false);
-
- virtual status_t setGain(float gain);
-
- virtual status_t standby();
-
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
-
- virtual unsigned int getInputFramesLost() const { return 0; }
-
- bool isActive() { return mPowerLock; }
-
- // ALSABufferProvider
- virtual status_t getNextBuffer(ALSABufferProvider::Buffer* buffer);
- virtual void releaseBuffer(ALSABufferProvider::Buffer* buffer);
-
- private:
- AudioHardwareALSA *mParent;
- bool mPowerLock;
- ALSADownsampler *mDownSampler;
- status_t mReadStatus;
- size_t mInPcmInBuf;
- int16_t *mPcmIn;
- };
-
- 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();
-
- /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
- virtual status_t setVoiceVolume(float volume);
-
- virtual status_t setMode(int mode);
-
- /**
- * 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);
-
- /** 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[];
- static uint32_t bufferRatio(uint32_t samplingRate);
-
- int mode() { return mMode; }
- Mutex& lock() { return mLock; }
-
- int setVoiceRecordGain(bool enable);
- int setVoiceRecordGain_l(bool enable);
-
- virtual status_t setParameters(const String8& keyValuePairs);
-
- 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.
- */
- status_t doRouting(uint32_t device, bool force = false);
- status_t doRouting_l(uint32_t device, bool force = false);
-
- virtual status_t dump(int fd, const Vector<String16>& args);
-
- friend class AudioStreamOutALSA;
- friend class AudioStreamInALSA;
-
- ALSAMixer *mMixer;
- AudioStreamOutALSA *mOutput;
- AudioStreamInALSA *mInput;
-
- private:
- Mutex mLock;
- void *mSecRilLibHandle;
- HRilClient mRilClient;
- bool mVrModeEnabled;
- bool mActivatedCP;
- bool mBluetoothECOff;
-
- HRilClient (*openClientRILD) (void);
- int (*disconnectRILD) (HRilClient);
- int (*closeClientRILD) (HRilClient);
- int (*isConnectedRILD) (HRilClient);
- int (*connectRILD) (HRilClient);
- int (*setCallVolume) (HRilClient, SoundType, int);
- int (*setCallAudioPath)(HRilClient, AudioPath);
- int (*setCallClockSync)(HRilClient, SoundClockCondition);
-
- void loadRILD(void);
- status_t connectRILDIfRequired(void);
- void setBluetoothNrEcOnOff(bool disable);
-
- };
-
- class ALSADownsampler {
- public:
- ALSADownsampler(uint32_t outSampleRate,
- uint32_t channelCount,
- uint32_t frameCount,
- ALSABufferProvider* provider);
-
- virtual ~ALSADownsampler();
-
- void reset();
- status_t initCheck() { return mStatus; }
- int resample(int16_t* out, size_t *outFrameCount);
-
- private:
- status_t mStatus;
- ALSABufferProvider* mProvider;
- uint32_t mSampleRate;
- uint32_t mChannelCount;
- uint32_t mFrameCount;
- int16_t *mInLeft;
- int16_t *mInRight;
- int16_t *mTmpLeft;
- int16_t *mTmpRight;
- int16_t *mTmp2Left;
- int16_t *mTmp2Right;
- int16_t *mOutLeft;
- int16_t *mOutRight;
- int mInInBuf;
- int mInTmpBuf;
- int mInTmp2Buf;
- int mOutBufPos;
- int mInOutBuf;
- };
-
-}; // namespace android
-#endif // ANDROID_AUDIO_HARDWARE_ALSA_H
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