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402 lines
13 KiB
402 lines
13 KiB
/*
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** Copyright 2008, The Android Open-Source Project
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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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/*
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**
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**
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**
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** Audio Hardware Commit log
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**
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**V1.0.0
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** 1)Merge from 4.4 and fix some compile error
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**
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*/
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//AudioHardware Version
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#define AUDIO_HAL_VERSION_NAME "sys.audio.version"
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#define AUDIO_HAL_VERSION "1.0.0"
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#ifndef ANDROID_AUDIO_HARDWARE_H
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#define ANDROID_AUDIO_HARDWARE_H
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#include <stdint.h>
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#include <sys/types.h>
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#include <utils/threads.h>
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#include <utils/SortedVector.h>
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#include <hardware_legacy/AudioHardwareBase.h>
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#include "secril-client.h"
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#include <speex/speex.h>
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#include <speex/speex_preprocess.h>
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#include <speex/speex_resampler.h>
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extern "C" {
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struct pcm;
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struct mixer;
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struct mixer_ctl;
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};
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namespace android_audio_legacy {
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// TODO: determine actual audio DSP and hardware latency
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// Additionnal latency introduced by audio DSP and hardware in ms
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#define AUDIO_HW_OUT_LATENCY_MS 0
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// Default audio output sample rate
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#define AUDIO_HW_OUT_SAMPLERATE 44100
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// Default audio output channel mask
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#define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO)
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// Default audio output sample format
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#define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT)
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// Kernel pcm out buffer size in frames at 44.1kHz
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#define AUDIO_HW_OUT_PERIOD_MULT 16 // (16 * 64 = 1024 frames)
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#define AUDIO_HW_OUT_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_OUT_PERIOD_MULT)
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#define AUDIO_HW_OUT_PERIOD_CNT 4
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// Default audio output buffer size in bytes
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#define AUDIO_HW_OUT_PERIOD_BYTES (AUDIO_HW_OUT_PERIOD_SZ * 2 * sizeof(int16_t))
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// Default audio input sample rate
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#define AUDIO_HW_IN_SAMPLERATE 44100
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// Default audio input channel mask
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#define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_STEREO)
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// Default audio input sample format
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#define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT)
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// Number of buffers in audio driver for input
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#define AUDIO_HW_NUM_IN_BUF 4
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// Kernel pcm in buffer size in frames at 44.1kHz (before resampling)
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#define AUDIO_HW_IN_PERIOD_MULT 16 // (8* 64 = 512 frames)
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#define AUDIO_HW_IN_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_IN_PERIOD_MULT)
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#define AUDIO_HW_IN_PERIOD_CNT 6
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// Default audio input buffer size in bytes (8kHz mono)
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#define AUDIO_HW_IN_PERIOD_BYTES ((AUDIO_HW_IN_PERIOD_SZ*sizeof(int16_t))/8)
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#define INPUT_SOURCE_KEY "Input Source"
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//1:Enable the AGC funtion ;0: disable the AGC function
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#define SPEEX_AGC_ENABLE 0
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//1:Enable the denoise funtion ;0: disable the denoise function
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#define SPEEX_DENOISE_ENABLE 1
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#define RESAMPLER_QUALITY SPEEX_RESAMPLER_QUALITY_DEFAULT
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class AudioHardware : public AudioHardwareBase
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{
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class AudioStreamOutALSA;
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class AudioStreamInALSA;
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public:
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AudioHardware();
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virtual ~AudioHardware();
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virtual status_t initCheck();
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virtual status_t setVoiceVolume(float volume);
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virtual status_t setMasterVolume(float volume);
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virtual status_t setMode(int mode);
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virtual status_t setMicMute(bool state);
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virtual status_t getMicMute(bool* state);
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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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virtual android_audio_legacy::AudioStreamOut* openOutputStream(
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uint32_t devices, int *format=0, uint32_t *channels=0,
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uint32_t *sampleRate=0, status_t *status=0);
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virtual android_audio_legacy::AudioStreamIn* openInputStream(
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uint32_t devices, int *format, uint32_t *channels,
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uint32_t *sampleRate, status_t *status,
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AudioSystem::audio_in_acoustics acoustics);
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virtual status_t setMasterMute(bool muted) ;
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virtual int createAudioPatch(unsigned int num_sources,
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const struct audio_port_config *sources,
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unsigned int num_sinks,
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const struct audio_port_config *sinks,
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audio_patch_handle_t *handle) ;
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virtual int releaseAudioPatch(audio_patch_handle_t handle) ;
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virtual int getAudioPort(struct audio_port *port) ;
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virtual int setAudioPortConfig(const struct audio_port_config *config) ;
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virtual android_audio_legacy::AudioStreamOut* openOutputStreamWithFlags(uint32_t devices,
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audio_output_flags_t flags,
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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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virtual void closeOutputStream(android_audio_legacy::AudioStreamOut* out);
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virtual void closeInputStream(android_audio_legacy::AudioStreamIn* in);
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virtual size_t getInputBufferSize(
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uint32_t sampleRate, int format, int channelCount);
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int mode() { return mMode; }
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unsigned getOutputRouteFromDevice(uint32_t device);
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unsigned getInputRouteFromDevice(uint32_t device);
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unsigned getVoiceRouteFromDevice(uint32_t device);
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unsigned getRouteFromDevice(uint32_t device);
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status_t setIncallPath_l(uint32_t device);
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status_t setInputSource_l(String8 source);
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static uint32_t getInputSampleRate(uint32_t sampleRate);
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android::sp <AudioStreamInALSA> getActiveInput_l();
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android::Mutex& lock() { return mLock; }
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struct pcm *openPcmOut_l();
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void closePcmOut_l();
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struct pcm *getPcm() { return mPcm; };
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android::sp <AudioStreamOutALSA> output() { return mOutput; }
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protected:
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virtual status_t dump(int fd, const Vector<String16>& args);
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private:
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bool mInit;
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bool mMicMute;
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android::sp <AudioStreamOutALSA> mOutput;
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android::SortedVector < android::sp<AudioStreamInALSA> > mInputs;
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android::Mutex mLock;
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struct pcm* mPcm;
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uint32_t mPcmOpenCnt;
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uint32_t mMixerOpenCnt;
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bool mInCallAudioMode;
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bool mVoipAudioMode;
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String8 mInputSource;
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bool mBluetoothNrec;
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void* mSecRilLibHandle;
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HRilClient mRilClient;
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bool mActivatedCP;
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HRilClient (*openClientRILD) (void);
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int (*disconnectRILD) (HRilClient);
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int (*closeClientRILD) (HRilClient);
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int (*isConnectedRILD) (HRilClient);
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int (*connectRILD) (HRilClient);
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int (*setCallVolume) (HRilClient, SoundType, int);
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int (*setCallAudioPath)(HRilClient, AudioPath);
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int (*setCallClockSync)(HRilClient, SoundClockCondition);
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void loadRILD(void);
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status_t connectRILDIfRequired(void);
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// trace driver operations for dump
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int mDriverOp;
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static uint32_t checkInputSampleRate(uint32_t sampleRate);
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static const uint32_t inputSamplingRates[];
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class AudioStreamOutALSA : public AudioStreamOut, public android::RefBase
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{
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public:
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AudioStreamOutALSA();
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virtual ~AudioStreamOutALSA();
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status_t set(AudioHardware* mHardware,
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uint32_t devices,
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int *pFormat,
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uint32_t *pChannels,
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uint32_t *pRate);
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virtual uint32_t sampleRate()
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const { return mSampleRate; }
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virtual size_t bufferSize()
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const { return mBufferSize; }
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virtual uint32_t channels()
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const { return mChannels; }
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virtual int format()
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const { return AUDIO_HW_OUT_FORMAT; }
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virtual uint32_t latency()
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const { return (1000 * AUDIO_HW_OUT_PERIOD_CNT *
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(bufferSize()/frameSize()))/sampleRate() +
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AUDIO_HW_OUT_LATENCY_MS; }
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virtual status_t setVolume(float left, float right)
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{ return INVALID_OPERATION; }
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virtual ssize_t write(const void* buffer, size_t bytes);
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virtual status_t standby();
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bool checkStandby();
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virtual status_t dump(int fd, const Vector<String16>& args);
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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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uint32_t device() { return mDevices; }
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virtual status_t getRenderPosition(uint32_t *dspFrames);
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void doStandby_l();
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void close_l();
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status_t open_l();
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int standbyCnt() { return mStandbyCnt; }
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void lock() { mLock.lock(); }
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void unlock() { mLock.unlock(); }
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private:
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android::Mutex mLock;
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AudioHardware* mHardware;
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struct mixer_ctl *mRouteCtl;
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const char *next_route;
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bool mStandby;
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uint32_t mDevices;
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uint32_t mChannels;
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uint32_t mSampleRate;
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size_t mBufferSize;
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// trace driver operations for dump
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int mDriverOp;
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int mStandbyCnt;
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};
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class DownSampler;
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class BufferProvider
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{
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public:
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struct Buffer {
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union {
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void* raw;
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short* i16;
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int8_t* i8;
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};
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size_t frameCount;
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};
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virtual ~BufferProvider() {}
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virtual status_t getNextBuffer(Buffer* buffer) = 0;
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virtual void releaseBuffer(Buffer* buffer) = 0;
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};
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class DownSampler {
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public:
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DownSampler(uint32_t outSampleRate,
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uint32_t inSampleRate,
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uint32_t channelCount,
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uint32_t frameCount,
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BufferProvider* provider);
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virtual ~DownSampler();
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void reset();
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status_t initCheck() { return mStatus; }
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int resample(int16_t* out, size_t *outFrameCount);
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private:
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status_t mStatus;
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BufferProvider* mProvider;
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uint32_t mSampleRate;
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uint32_t mChannelCount;
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uint32_t mFrameCount;
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int16_t *mTmpOutBuf;
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int mOutBufPos;
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int mInOutBuf;
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int mInInBuf;
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SpeexResamplerState *mInResampler; // handle on input speex resampler
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};
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class AudioStreamInALSA : public AudioStreamIn, public BufferProvider, public android::RefBase
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{
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public:
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AudioStreamInALSA();
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virtual ~AudioStreamInALSA();
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status_t set(AudioHardware* hw,
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uint32_t devices,
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int *pFormat,
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uint32_t *pChannels,
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uint32_t *pRate,
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AudioSystem::audio_in_acoustics acoustics);
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virtual size_t bufferSize() const { return mBufferSize; }
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virtual uint32_t channels() const { return mChannels; }
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virtual int format() const { return AUDIO_HW_IN_FORMAT; }
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virtual uint32_t sampleRate() const { return mSampleRate; }
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virtual status_t setGain(float gain);// { return INVALID_OPERATION; }
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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 standby();
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bool checkStandby();
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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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virtual unsigned int getInputFramesLost() const { return 0; }
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uint32_t device() { return mDevices; }
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void doStandby_l();
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void close_l();
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status_t open_l();
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int standbyCnt() { return mStandbyCnt; }
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static size_t getBufferSize(uint32_t sampleRate, int channelCount);
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// BufferProvider
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virtual status_t getNextBuffer(BufferProvider::Buffer* buffer);
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virtual void releaseBuffer(BufferProvider::Buffer* buffer);
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void lock() { mLock.lock(); }
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void unlock() { mLock.unlock(); }
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virtual status_t addAudioEffect(effect_handle_t effect){return 0;};
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virtual status_t removeAudioEffect(effect_handle_t effect){return 0;};
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private:
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android::Mutex mLock;
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AudioHardware* mHardware;
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struct pcm *mPcm;
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struct mixer_ctl *mRouteCtl;
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const char *next_route;
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bool mStandby;
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uint32_t mDevices;
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uint32_t mChannels;
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uint32_t mChannelCount;
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uint32_t mSampleRate;
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uint32_t mReqSampleRate;
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uint32_t mInSampleRate;
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size_t mBufferSize;
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DownSampler *mDownSampler;
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status_t mReadStatus;
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size_t mInPcmInBuf;
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int16_t *mPcmIn;
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bool mMicMute;
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// trace driver operations for dump
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int mDriverOp;
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int mStandbyCnt;
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uint32_t mDropCnt;
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#if (SPEEX_AGC_ENABLE||SPEEX_DENOISE_ENABLE)
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SpeexPreprocessState* mSpeexState;
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int mSpeexFrameSize;
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int16_t *mSpeexPcmIn;
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#endif//SPEEX_AGC_ENABLE||SPEEX_DENOISE_ENABLE
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};
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};
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}; // namespace android
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#endif
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