380 lines
11 KiB
C
380 lines
11 KiB
C
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/* AudioHardwareALSA.h
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**
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** Copyright 2008-2009, Wind River Systems
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**
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** Licensed under the Apache License, Version 2.0 (the "License");
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** you may not use this file except in compliance with the License.
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** You may obtain a copy of the License at
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**
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** http://www.apache.org/licenses/LICENSE-2.0
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**
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** Unless required by applicable law or agreed to in writing, software
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** distributed under the License is distributed on an "AS IS" BASIS,
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** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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** See the License for the specific language governing permissions and
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** limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_HARDWARE_ALSA_H
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#define ANDROID_AUDIO_HARDWARE_ALSA_H
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#include <utils/List.h>
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#include <hardware_legacy/AudioHardwareBase.h>
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#include <alsa/asoundlib.h>
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#include <hardware/hardware.h>
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namespace android
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{
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class AudioHardwareALSA;
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/**
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* The id of ALSA module
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*/
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#define ALSA_HARDWARE_MODULE_ID "alsa"
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#define ALSA_HARDWARE_NAME "alsa"
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struct alsa_device_t;
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struct alsa_handle_t {
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alsa_device_t * module;
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uint32_t devices;
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uint32_t curDev;
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int curMode;
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snd_pcm_t * handle;
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snd_pcm_format_t format;
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uint32_t channels;
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uint32_t sampleRate;
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unsigned int latency; // Delay in usec
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unsigned int bufferSize; // Size of sample buffer
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int mmap;
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void * modPrivate;
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};
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typedef List<alsa_handle_t> ALSAHandleList;
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struct alsa_device_t {
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hw_device_t common;
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status_t (*init)(alsa_device_t *, ALSAHandleList &);
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status_t (*open)(alsa_handle_t *, uint32_t, int);
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status_t (*close)(alsa_handle_t *);
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status_t (*standby)(alsa_handle_t *);
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status_t (*route)(alsa_handle_t *, uint32_t, int);
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status_t (*voicevolume)(float);
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status_t (*set)(const String8&);
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status_t (*resetDefaults)(alsa_handle_t *handle);
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};
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/**
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* The id of acoustics module
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*/
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#define ACOUSTICS_HARDWARE_MODULE_ID "acoustics"
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#define ACOUSTICS_HARDWARE_NAME "acoustics"
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struct acoustic_device_t {
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hw_device_t common;
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// Required methods...
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status_t (*use_handle)(acoustic_device_t *, alsa_handle_t *);
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status_t (*cleanup)(acoustic_device_t *);
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status_t (*set_params)(acoustic_device_t *, AudioSystem::audio_in_acoustics, void *);
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// Optional methods...
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ssize_t (*read)(acoustic_device_t *, void *, size_t);
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ssize_t (*write)(acoustic_device_t *, const void *, size_t);
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status_t (*recover)(acoustic_device_t *, int);
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void * modPrivate;
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};
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// ----------------------------------------------------------------------------
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class ALSAMixer
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{
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public:
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ALSAMixer();
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virtual ~ALSAMixer();
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bool isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
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status_t setMasterVolume(float volume);
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status_t setMasterGain(float gain);
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status_t setVolume(uint32_t device, float left, float right);
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status_t setGain(uint32_t device, float gain);
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status_t setCaptureMuteState(uint32_t device, bool state);
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status_t getCaptureMuteState(uint32_t device, bool *state);
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status_t setPlaybackMuteState(uint32_t device, bool state);
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status_t getPlaybackMuteState(uint32_t device, bool *state);
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private:
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snd_mixer_t * mMixer[SND_PCM_STREAM_LAST+1];
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};
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class ALSAControl
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{
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public:
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ALSAControl(const char *device = "hw:00");
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virtual ~ALSAControl();
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status_t get(const char *name, unsigned int &value, int index = 0);
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status_t set(const char *name, unsigned int value, int index = -1);
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status_t set(const char *name, const char *);
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status_t getmin(const char *name, unsigned int &max);
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status_t getmax(const char *name, unsigned int &min);
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private:
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snd_ctl_t * mHandle;
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};
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class ALSAStreamOps
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{
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public:
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ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle);
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virtual ~ALSAStreamOps();
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status_t set(int *format, uint32_t *channels, uint32_t *rate);
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status_t setParameters(const String8& keyValuePairs);
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String8 getParameters(const String8& keys);
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uint32_t sampleRate() const;
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size_t bufferSize() const;
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int format() const;
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uint32_t channels() const;
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status_t open(int mode);
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void close();
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protected:
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friend class AudioHardwareALSA;
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acoustic_device_t *acoustics();
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ALSAMixer *mixer();
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AudioHardwareALSA * mParent;
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alsa_handle_t * mHandle;
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Mutex mLock;
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bool mPowerLock;
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};
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// ----------------------------------------------------------------------------
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class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
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{
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public:
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AudioStreamOutALSA(AudioHardwareALSA *parent, alsa_handle_t *handle);
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virtual ~AudioStreamOutALSA();
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virtual uint32_t sampleRate() const
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{
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return ALSAStreamOps::sampleRate();
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}
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virtual size_t bufferSize() const
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{
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return ALSAStreamOps::bufferSize();
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}
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virtual uint32_t channels() const;
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virtual int format() const
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{
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return ALSAStreamOps::format();
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}
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virtual uint32_t latency() const;
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virtual ssize_t write(const void *buffer, size_t bytes);
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virtual status_t dump(int fd, const Vector<String16>& args);
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status_t setVolume(float left, float right);
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virtual status_t standby();
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virtual status_t setParameters(const String8& keyValuePairs) {
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return ALSAStreamOps::setParameters(keyValuePairs);
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}
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virtual String8 getParameters(const String8& keys) {
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return ALSAStreamOps::getParameters(keys);
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}
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// return the number of audio frames written by the audio dsp to DAC since
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// the output has exited standby
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virtual status_t getRenderPosition(uint32_t *dspFrames);
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status_t open(int mode);
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status_t close();
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private:
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uint32_t mFrameCount;
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};
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class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
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{
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public:
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AudioStreamInALSA(AudioHardwareALSA *parent,
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alsa_handle_t *handle,
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AudioSystem::audio_in_acoustics audio_acoustics);
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virtual ~AudioStreamInALSA();
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virtual uint32_t sampleRate() const
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{
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return ALSAStreamOps::sampleRate();
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}
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virtual size_t bufferSize() const
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{
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return ALSAStreamOps::bufferSize();
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}
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virtual uint32_t channels() const
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{
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return ALSAStreamOps::channels();
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}
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virtual int format() const
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{
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return ALSAStreamOps::format();
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}
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virtual ssize_t read(void* buffer, ssize_t bytes);
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virtual status_t dump(int fd, const Vector<String16>& args);
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virtual status_t setGain(float gain);
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virtual status_t standby();
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virtual status_t setParameters(const String8& keyValuePairs)
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{
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return ALSAStreamOps::setParameters(keyValuePairs);
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}
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virtual String8 getParameters(const String8& keys)
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{
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return ALSAStreamOps::getParameters(keys);
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}
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// Return the amount of input frames lost in the audio driver since the last call of this function.
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// Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
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// Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
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// Unit: the number of input audio frames
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virtual unsigned int getInputFramesLost() const;
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status_t setAcousticParams(void* params);
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status_t open(int mode);
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status_t close();
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private:
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void resetFramesLost();
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unsigned int mFramesLost;
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AudioSystem::audio_in_acoustics mAcoustics;
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};
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class AudioHardwareALSA : public AudioHardwareBase
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{
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public:
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AudioHardwareALSA();
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virtual ~AudioHardwareALSA();
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/**
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* check to see if the audio hardware interface has been initialized.
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* return status based on values defined in include/utils/Errors.h
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*/
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virtual status_t initCheck();
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/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
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virtual status_t setVoiceVolume(float volume);
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/**
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* set the audio volume for all audio activities other than voice call.
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* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
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* the software mixer will emulate this capability.
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*/
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virtual status_t setMasterVolume(float volume);
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/**
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* setMode is called when the audio mode changes. NORMAL mode is for
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* standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
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* when a call is in progress.
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*/
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virtual status_t setMode(int mode);
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// mic mute
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virtual status_t setMicMute(bool state);
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virtual status_t getMicMute(bool* state);
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// set/get global audio parameters
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virtual status_t setParameters(const String8& keyValuePairs);
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//virtual String8 getParameters(const String8& keys);
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// Returns audio input buffer size according to parameters passed or 0 if one of the
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// parameters is not supported
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virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channels);
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/** This method creates and opens the audio hardware output stream */
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virtual AudioStreamOut* openOutputStream(
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uint32_t devices,
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int *format=0,
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uint32_t *channels=0,
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uint32_t *sampleRate=0,
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status_t *status=0);
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virtual void closeOutputStream(AudioStreamOut* out);
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/** This method creates and opens the audio hardware input stream */
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virtual AudioStreamIn* openInputStream(
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uint32_t devices,
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int *format,
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uint32_t *channels,
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uint32_t *sampleRate,
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status_t *status,
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AudioSystem::audio_in_acoustics acoustics);
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virtual void closeInputStream(AudioStreamIn* in);
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/**This method dumps the state of the audio hardware */
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//virtual status_t dumpState(int fd, const Vector<String16>& args);
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static AudioHardwareInterface* create();
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int mode()
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{
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return mMode;
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}
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protected:
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virtual status_t dump(int fd, const Vector<String16>& args);
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friend class AudioStreamOutALSA;
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friend class AudioStreamInALSA;
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friend class ALSAStreamOps;
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ALSAMixer * mMixer;
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alsa_device_t * mALSADevice;
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acoustic_device_t * mAcousticDevice;
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ALSAHandleList mDeviceList;
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private:
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Mutex mLock;
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};
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// ----------------------------------------------------------------------------
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}; // namespace android
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#endif // ANDROID_AUDIO_HARDWARE_ALSA_H
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