Android-NDK-audio-echo

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原创最新推荐文章于 2026-10-13 15:04:37 发布 761 阅读 AI 写同款· GEO 优化 ›

Unity安卓打包失败根源：JDK/SDK/NDK三重环境链深度解析 Android SDK和JDK是安卓应用开发的基础技术栈，前者提供aapt、adb、d8等原生构建工具，后者负责Java字节码编译与APK签名；二者通过NDK延伸至C++层，共同支撑Unity的IL2CPP后端。其核心原理在于版本强约束与路径严格校验——JDK字节码版本不匹配会触发major version错误，SDK组件缺失将导致aapt2或zipalign命令未找到，NDK路径指向子目录而非根目录则无法读取source.properties。这种精密依赖构成工程落地的关键技术门槛，广泛应用于Unity游阅读详情

项目图

运行界面

界面分析

setContentView(R.layout.activity_main);//设置布局文件

controlButton = (Button)findViewById((R.id.capture_control_button));


statusView = (TextView)findViewById(R.id.statusView);

{

private void queryNativeAudioParameters() {
    supportRecording = true;
    AudioManager myAudioMgr = (AudioManager) getSystemService(Context.AUDIO_SERVICE);//获取audioManager的服务
    if(myAudioMgr == null) {
        supportRecording = false;
        return;
    }
    nativeSampleRate  =  myAudioMgr.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);//audio的采样率
    nativeSampleBufSize =myAudioMgr.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);//每个sample的大小
{

音频源:我们可以使用麦克风作为采集音频的数据源。
采样率:一秒钟对声音数据的采样次数，采样率越高，音质越好。
音频通道：单声道，双声道等，
音频格式:一般选用PCM格式，即原始的音频样本。
缓冲区大小:音频数据写入缓冲区的总数，可以通过AudioRecord.getMinBufferSize获取最小的缓冲区。（将音频采集到缓冲区中然后再从缓冲区中读取）

}

    // hardcoded channel to mono: both sides -- C++ and Java sides
    int recBufSize = AudioRecord.getMinBufferSize(
            Integer.parseInt(nativeSampleRate),
            AudioFormat.CHANNEL_IN_MONO,
            AudioFormat.ENCODING_PCM_16BIT);
    if (recBufSize == AudioRecord.ERROR ||
            recBufSize == AudioRecord.ERROR_BAD_VALUE) {
        supportRecording = false;
    }

}

}

queryNativeAudioParameters();

delaySeekBar = (SeekBar)findViewById(R.id.delaySeekBar);
curDelayTV = (TextView)findViewById(R.id.curDelay);
echoDelayProgress = delaySeekBar.getProgress() * 1000 / delaySeekBar.getMax();

设置seekbar的监听器

delaySeekBar.setOnSeekBarChangeListener(new SeekBar.OnSeekBarChangeListener() {
    @Override
    public void onProgressChanged(SeekBar seekBar, int progress, boolean fromUser) {
        float curVal = (float)progress / delaySeekBar.getMax();
        curDelayTV.setText(String.format("%s", curVal));
{

//设置坐标

private void setSeekBarPromptPosition(SeekBar seekBar, TextView label) {
    float thumbX = (float)seekBar.getProgress()/ seekBar.getMax() *
                          seekBar.getWidth() + seekBar.getX();
    label.setX(thumbX - label.getWidth()/2.0f);
}

}
        setSeekBarPromptPosition(delaySeekBar, curDelayTV);
        if (!fromUser) return;

        echoDelayProgress = progress * 1000 / delaySeekBar.getMax();
{

static native boolean configureEcho(int delayInMs, float decay);

}
        configureEcho(echoDelayProgress, echoDecayProgress);
    }
    @Override
    public void onStartTrackingTouch(SeekBar seekBar) {}
    @Override
    public void onStopTrackingTouch(SeekBar seekBar) {}
});

//当前view 在attachedToWindow之后执行操作
delaySeekBar.post(new Runnable() {
    @Override
    public void run() {
        setSeekBarPromptPosition(delaySeekBar, curDelayTV);
    }
});

private void startEcho() {
    if(!supportRecording){
        return;
    }
    if (!isPlaying) {
        if(!createSLBufferQueueAudioPlayer()) {
            statusView.setText(getString(R.string.player_error_msg));
            return;
        }
        if(!createAudioRecorder()) {
            deleteSLBufferQueueAudioPlayer();
            statusView.setText(getString(R.string.recorder_error_msg));
            return;
        }
        startPlay();   // startPlay() triggers startRecording()
        statusView.setText(getString(R.string.echoing_status_msg));
    } else {
        stopPlay();  // stopPlay() triggers stopRecording()
        updateNativeAudioUI();
        deleteAudioRecorder();
        deleteSLBufferQueueAudioPlayer();
    }
    isPlaying = !isPlaying;
    controlButton.setText(getString(isPlaying ?
            R.string.cmd_stop_echo: R.string.cmd_start_echo));
}

jni function 声明

/*
 * jni function declarations
 */
static native void createSLEngine(int rate, int framesPerBuf,
                                  long delayInMs, float decay);
static native void deleteSLEngine();
static native boolean configureEcho(int delayInMs, float decay);
static native boolean createSLBufferQueueAudioPlayer();
static native void deleteSLBufferQueueAudioPlayer();

static native boolean createAudioRecorder();
static native void deleteAudioRecorder();
static native void startPlay();
static native void stopPlay();

OpenSL ES

以上图片引用自：https://www.jianshu.com/p/82da5f87314f

OpensSL ES是无授权费，跨平台的，针对嵌入式精心优化的硬件音频加速API。

对象和接口的概念：

对象：提供一组资源及其状态的抽象

接口：提供特定功能的方法的抽象

对象与接口的关系：

对象暴露的接口，有以下三个方面决定：

1）对象的类型

2）应用程序在对象创建期间的，接口请求。

3）在对象声明周期，接口的添加和移除。

一个对象的类型，表明了它有implicat interfaces，implicat interface的含义是：无论应用程序是否request，它都存在对象暴露的接口函数中。

在对象创建的时候，如果有应用请求才暴露的接口，被称为explicit interfaces。

对象定义的可以动态添加和移除的接口，被称为dynamic interfaces.SLDynamicInterfaceManagementItf

EchoAudioEngine

struct EchoAudioEngine {
  SLmilliHertz fastPathSampleRate_;
  uint32_t fastPathFramesPerBuf_;
  uint16_t sampleChannels_;
  uint16_t bitsPerSample_;

  SLObjectItf slEngineObj_;
  SLEngineItf slEngineItf_;

  AudioRecorder *recorder_;
  AudioPlayer *player_;
  AudioQueue *freeBufQueue_;  // Owner of the queue
  AudioQueue *recBufQueue_;   // Owner of the queue

  sample_buf *bufs_;
  uint32_t bufCount_;
  uint32_t frameCount_;
  int64_t echoDelay_;
  float echoDecay_;
  AudioDelay *delayEffect_;
};
static EchoAudioEngine engine;

SLObjectItf:任何对象都暴露这个接口。每个方法创建一个对象，都返回这个接口SLObjectItf. 销毁对象通过destory（）.应用程序获取其他的接口，通过type ID 使用GetInterface来返回。通过SLObjectItf接口的Realize和resume来控制状态。

SLEngineItf:应用程序开启一个回话的方式，是通过创建一个engine对象的。Engine对象的创建是通过一个SLCreateEngine()来获取的，返回一个SLObjectItf.

当Engine对象创建后，可以获取它的SLEngineItf。

SLBufferQueueItf：被用于流式的音频数据，填充到一个player object或者record object的buffer队列里面。

1）对于recorder对象，当recorder的状态处于SL_RECORDSTATE_RECORDING时。这个对象被SLRecordItf接口控制添加buffer，来隐含开始填充的进程。如果队列中没有足够的buffer，这个auido的数据的填充将会停止和在buffer队列中的要被录制的audio数据会丢失。这个录制仍旧是SL_RECORDSTATE_RECORDING状态。一旦入队了额外的buffer，

填充音频数据将与当前音频数据一起恢复，而不是从饥饿开始了。如果recorder没有处于录制状态，额外的buffer并没有填充队列中任何buffer.

2)在播放对象中的buffer被就地使用，并不会被设备拷贝。应用程序的开发者应该注意到，修改在已经入队的buffer的内容是无效的和会引起音频数据损坏的。

3）一旦入队列的buffer完成了播放或者填充，有callback进行通知，它是安全删除buffer。它也是安全的用新数据填充buffer，和在在播放对象入队buffer和再次入队buffer在录制对象。

4）状态转换为SL_PLAYSTATE_STOPPED，通过release所有的buffer来清空队列，和设置cursor为0.每一个buffer被释放，都会回调被带有SL_BUFFERQUEUENVENT_STOP的flag.

5)一旦转换为了SL_RECORDSTATE_STOPPED状态，应用程序应该继续放buffer到队列中，来取回系统中剩余的buffer。获取剩余的buffer完成的标志是回调方法带有了SL_BUFFERQUEUEEVNENT_CONTEN_END的事件flag。空的buffer可以被用于下一个录制的回话。录制的cursor被设置为0.

6）一旦转化为SL_PLAYSTATE_PAUSEDor SL_RECORDSTATE_PAUSED，cursor仍旧保留在当前的位置。

createSLEngine

engine.fastPathSampleRate_ = static_cast<SLmilliHertz>(sampleRate) * 1000;
engine.fastPathFramesPerBuf_ = static_cast<uint32_t>(framesPerBuf);
engine.sampleChannels_ = AUDIO_SAMPLE_CHANNELS;
engine.bitsPerSample_ = SL_PCMSAMPLEFORMAT_FIXED_16;
/*
SL_API SLresultSLAPIENTRY slCreateEngine(
SLObjectItf *pEngine,
SLuint32 numOptions
const SLEngineOption *pEngineOptions,
SLuint32 numInterfaces,
const SLInterfaceID *pInterfaceIds,
const SLboolean * pInterfaceRequired
)


*/


result = slCreateEngine(&engine.slEngineObj_, 0, NULL, 0, NULL, NULL);
SLASSERT(result);

//Realizing the object in synchronous mode. */

result =
    (*engine.slEngineObj_)->Realize(engine.slEngineObj_, SL_BOOLEAN_FALSE);
SLASSERT(result);

//获取slEngineItf_

result = (*engine.slEngineObj_)
             ->GetInterface(engine.slEngineObj_, SL_IID_ENGINE,
                            &engine.slEngineItf_);

// compute the RECOMMENDED fast audio buffer size:
//   the lower latency required
//     *) the smaller the buffer should be (adjust it here) AND
//     *) the less buffering should be before starting player AFTER
//        receiving the recorder buffer
//   Adjust the bufSize here to fit your bill [before it busts]
uint32_t bufSize = engine.fastPathFramesPerBuf_ * engine.sampleChannels_ *
                   engine.bitsPerSample_;
bufSize = (bufSize + 7) >> 3;  // bits --> byte
engine.bufCount_ = BUF_COUNT;
engine.bufs_ = allocateSampleBufs(engine.bufCount_, bufSize);
assert(engine.bufs_);

//创建缓冲队列,freeBufQueue是指空闲的buffer队列，主要是提供空的采样数组。recBufQueue是接收缓冲队列，主要是用来存储已采集到的音频数据，同样也是播放数据的来源。引擎初始化完毕之后会初始化freeBufQueue，初始化了16个空的大小为480字节的数组。至此音频引擎的初始化结束。

engine.freeBufQueue_ = new AudioQueue(engine.bufCount_);
engine.recBufQueue_ = new AudioQueue(engine.bufCount_);
assert(engine.freeBufQueue_ && engine.recBufQueue_);
for (uint32_t i = 0; i < engine.bufCount_; i++) {
  engine.freeBufQueue_->push(&engine.bufs_[i]);
}

//创建AudioDelay

engine.echoDelay_ = delayInMs;
engine.echoDecay_ = decay;
engine.delayEffect_ = new AudioDelay(
    engine.fastPathSampleRate_, .sampleChannels_, engine.bitsPerSample_,
    engine.echoDelay_, engine.echoDecay_);
assert(engine.delayEffect_);

AudioPlayer

构造方法

SLresult result;
assert(sampleFormat);
sampleInfo_ = *sampleFormat;
/*
SLresult (*CreateOutputMix) (
SLEngineItf self,
SLObjectItf* pMix,
SLuint32 numInterfaces,
const SLInterfaceID * pInterfaceIds,
const SLboolean * pInterfaceRequired
);
创建混音器的对象
*/
result = (*slEngine)
             ->CreateOutputMix(slEngine, &outputMixObjectItf_, 0, NULL, NULL);
SLASSERT(result);

// realize the output mix
result =
    (*outputMixObjectItf_)->Realize(outputMixObjectItf_, SL_BOOLEAN_FALSE);
SLASSERT(result);

// configure audio source，配置audio source
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
    SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, DEVICE_SHADOW_BUFFER_QUEUE_LEN};
SLAndroidDataFormat_PCM_EX format_pcm;
ConvertToSLSampleFormat(&format_pcm, &sampleInfo_);
SLDataSource audioSrc = {&loc_bufq, &format_pcm};

// configure audio sink,配置音频的输出
SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX,
                                      outputMixObjectItf_};
SLDataSink audioSnk = {&loc_outmix, NULL};

/*
 * create fast path audio player: SL_IID_BUFFERQUEUE and SL_IID_VOLUME
 * and other non-signal processing interfaces are ok.，创建audioplayer
 */
SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
/*
SLresult (*CreateAudioPlayer) (
SLEngineItf self,
SLObjectItf* pPlayer,
const SLDataSource *pAudioSrc,
const SLDataSink *pAudioSnk,
SLuint32 numInterfaces,
const SLInterfaceID * pInterfaceIds,
const SLboolean * pInterfaceRequired
);
创建音频播放器
*/
result = (*slEngine)->CreateAudioPlayer(
    slEngine, &playerObjectItf_, &audioSrc, &audioSnk,
    sizeof(ids) / sizeof(ids[0]), ids, req);
SLASSERT(result);

// realize the player，实现播放器
result = (*playerObjectItf_)->Realize(playerObjectItf_, SL_BOOLEAN_FALSE);
SLASSERT(result);

// get the play interface，获取播放接口
result = (*playerObjectItf_)
             ->GetInterface(playerObjectItf_, SL_IID_PLAY, &playItf_);
SLASSERT(result);

// get the buffer queue interface，获取bufferq queue的接口
result = (*playerObjectItf_)
             ->GetInterface(playerObjectItf_, SL_IID_BUFFERQUEUE,
                            &playBufferQueueItf_);
SLASSERT(result);

// register callback on the buffer queue，在bufferq queue上注册接口
result = (*playBufferQueueItf_)
             ->RegisterCallback(playBufferQueueItf_, bqPlayerCallback, this);
SLASSERT(result);

//设置播放状态

result = (*playItf_)->SetPlayState(playItf_, SL_PLAYSTATE_STOPPED);
SLASSERT(result);

// create an empty queue to track deviceQueue
devShadowQueue_ = new AudioQueue(DEVICE_SHADOW_BUFFER_QUEUE_LEN);
assert(devShadowQueue_);

silentBuf_.cap_ = (format_pcm.containerSize >> 3) * format_pcm.numChannels *
                  sampleInfo_.framesPerBuf_;
silentBuf_.buf_ = new uint8_t[silentBuf_.cap_];
memset(silentBuf_.buf_, 0, silentBuf_.cap_);
silentBuf_.size_ = silentBuf_.cap_;

AudioPlayer::Start

//首先获取播放状态

SLuint32 state;
SLresult result = (*playItf_)->GetPlayState(playItf_, &state);
if (result != SL_RESULT_SUCCESS) {
  return SL_BOOLEAN_FALSE;
}
if (state == SL_PLAYSTATE_PLAYING) {
  return SL_BOOLEAN_TRUE;
}
//先设置播放状态是STOPPED状态
result = (*playItf_)->SetPlayState(playItf_, SL_PLAYSTATE_STOPPED);
SLASSERT(result);
//然后如队列相应的buffer。
result =
    (*playBufferQueueItf_)
        ->Enqueue(playBufferQueueItf_, silentBuf_.buf_, silentBuf_.size_);
SLASSERT(result);
devShadowQueue_->push(&silentBuf_);
//设置播放的状态为playing
result = (*playItf_)->SetPlayState(playItf_, SL_PLAYSTATE_PLAYING);
SLASSERT(result);
return SL_BOOLEAN_TRUE;

ProcessSLCallback

注册callback：

void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *ctx) {
  (static_cast<AudioPlayer *>(ctx))->ProcessSLCallback(bq);
}

void AudioPlayer::ProcessSLCallback(SLAndroidSimpleBufferQueueItf bq) {
#ifdef ENABLE_LOG
  logFile_->logTime();
#endif
  std::lock_guard<std::mutex> lock(stopMutex_);

  // retrieve the finished device buf and put onto the free queue
  // so recorder could re-use it，获取已经完成的device buf，并把它放到空闲队列的缓冲区。recorder可以重复使用它
  sample_buf *buf;

devShadowQueue_->pop();

  if (buf != &silentBuf_) {
    buf->size_ = 0;
    freeQueue_->push(buf);

    if (!playQueue_->front(&buf)) {
#ifdef ENABLE_LOG
      logFile_->log("%s", "====Warning: running out of the Audio buffers");
#endif
      return;
    }

devShadowQueue_->push(buf);
(*bq)->Enqueue(bq, buf->buf_, buf->size_);//在从播放队列中，获取一个buffer并入队列。
playQueue_->pop();

//如果播放队列的大小小于kickstart buffer的数量。入队列silent数据。

playQueue是播放队列，如果为空的话表示没有缓冲数据，这里回调到用的地方做错误处理，若是成功取出，那么先将其存入中转队列，并且将其传入调用播放的方法中开启播放，最后在播放队列中删除该已经播放的数组，在播放完成之后会进入Player播放队列注册的回调中。

if (playQueue_->size() < PLAY_KICKSTART_BUFFER_COUNT) {
  (*bq)->Enqueue(bq, buf->buf_, buf->size_);
  devShadowQueue_->push(&silentBuf_);
  return;
}

//填充要播放的数据。

for (int32_t idx = 0; idx < PLAY_KICKSTART_BUFFER_COUNT; idx++) {
  playQueue_->front(&buf);
  playQueue_->pop();
  devShadowQueue_->push(buf);//devshadow的queue的作用是中转队列。
  (*bq)->Enqueue(bq, buf->buf_, buf->size_);
}

AudioPlayer::stop

SLuint32 state;

  SLresult result = (*playItf_)->GetPlayState(playItf_, &state);
  SLASSERT(result);

  if (state == SL_PLAYSTATE_STOPPED) return;

  std::lock_guard<std::mutex> lock(stopMutex_);

  result = (*playItf_)->SetPlayState(playItf_, SL_PLAYSTATE_STOPPED);
  SLASSERT(result);
  (*playBufferQueueItf_)->Clear(playBufferQueueItf_);

#ifdef ENABLE_LOG
  if (logFile_) {
    delete logFile_;
    logFile_ = nullptr;
  }
#endif

AudioPlayer:~AudioPlayer

AudioPlayer::~AudioPlayer() {
  std::lock_guard<std::mutex> lock(stopMutex_);

  // destroy buffer queue audio player object, and invalidate all associated
  // interfaces
  if (playerObjectItf_ != NULL) {
    (*playerObjectItf_)->Destroy(playerObjectItf_);
  }
  // Consume all non-completed audio buffers,消耗掉所有未完成的buffer
  sample_buf *buf = NULL;
  while (devShadowQueue_->front(&buf)) {
    buf->size_ = 0;
    devShadowQueue_->pop();
    if(buf != &silentBuf_) {
      freeQueue_->push(buf);
    }
  }
  delete devShadowQueue_;

//把正在播放的队列，放audiobuffer到freeQueue
  while (playQueue_->front(&buf)) {
    buf->size_ = 0;
    playQueue_->pop();
    freeQueue_->push(buf);
  }
//销毁混音器的对象接口
  // destroy output mix object, and invalidate all associated interfaces
  if (outputMixObjectItf_) {
    (*outputMixObjectItf_)->Destroy(outputMixObjectItf_);
  }

  delete[] silentBuf_.buf_;
}

AudioRecorder

AudioRecorder::AudioRecorder

//转为SL的格式

SLresult result;
sampleInfo_ = *sampleFormat;
SLAndroidDataFormat_PCM_EX format_pcm;
ConvertToSLSampleFormat(&format_pcm, &sampleInfo_);

// configure audio source,配置audiosource，SL_DATALOCATOR_IODEVICE Data will be generated or consumed by the
specified IO device. Note: for audio output use
the output mix.
SL_DEFAULTDEVICEID_AUDIOINPUTIdentifier denoting the set of input devicesfrom
whichthe implementation receives audio from by
default.

SLDataLocator_IODevice loc_dev = {SL_DATALOCATOR_IODEVICE, SL_IODEVICE_AUDIOINPUT, SL_DEFAULTDEVICEID_AUDIOINPUT, NULL}; SLDataSource audioSrc = {&loc_dev, NULL};

// configure audio sink，配置audi sink
SLDataLocator_AndroidSimpleBufferQueue loc_bq = {
    SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, DEVICE_SHADOW_BUFFER_QUEUE_LEN};

SLDataSink audioSnk = {&loc_bq, &format_pcm};

//创建audiorecorder的object

// create audio recorder
// (requires the RECORD_AUDIO permission)
const SLInterfaceID id[2] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
                             SL_IID_ANDROIDCONFIGURATION};
const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result = (*slEngine)->CreateAudioRecorder(
    slEngine, &recObjectItf_, &audioSrc, &audioSnk,
    sizeof(id) / sizeof(id[0]), id, req);
SLASSERT(result);

// Configure the voice recognition preset which has no
// signal processing for lower latency.
/*

/*---------------------------------------------------------------------------*/
/* Android AudioRecorder configuration                                       */
/*---------------------------------------------------------------------------*/
 
/** Audio recording preset */
/** Audio recording preset key */
#define SL_ANDROID_KEY_RECORDING_PRESET ((const SLchar*) "androidRecordingPreset")
/** Audio recording preset values */
/**   preset "none" cannot be set, it is used to indicate the current settings
 *     do not match any of the presets. */
#define SL_ANDROID_RECORDING_PRESET_NONE                ((SLuint32) 0x00000000)
/**   generic recording configuration on the platform */
#define SL_ANDROID_RECORDING_PRESET_GENERIC             ((SLuint32) 0x00000001)
/**   uses the microphone audio source with the same orientation as the camera
 *     if available, the main device microphone otherwise */
#define SL_ANDROID_RECORDING_PRESET_CAMCORDER           ((SLuint32) 0x00000002)
/**   uses the main microphone tuned for voice recognition */
#define SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION   ((SLuint32) 0x00000003)
/**   uses the main microphone tuned for audio communications */
#define SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION ((SLuint32) 0x00000004)
/**   uses the main microphone unprocessed */
#define SL_ANDROID_RECORDING_PRESET_UNPROCESSED         ((SLuint32) 0x00000005)
 
/*---------------------------------------------------------------------------*/
/* Android AudioPlayer configuration                                         */
/*---------------------------------------------------------------------------*/
 
/** Audio playback stream type */
/** Audio playback stream type key */
#define SL_ANDROID_KEY_STREAM_TYPE ((const SLchar*) "androidPlaybackStreamType")
 
/** Audio playback stream type  values */
/*      same as android.media.AudioManager.STREAM_VOICE_CALL */
#define SL_ANDROID_STREAM_VOICE        ((SLint32) 0x00000000)
/*      same as android.media.AudioManager.STREAM_SYSTEM */
#define SL_ANDROID_STREAM_SYSTEM       ((SLint32) 0x00000001)
/*      same as android.media.AudioManager.STREAM_RING */
#define SL_ANDROID_STREAM_RING         ((SLint32) 0x00000002)
/*      same as android.media.AudioManager.STREAM_MUSIC */
#define SL_ANDROID_STREAM_MEDIA        ((SLint32) 0x00000003)
/*      same as android.media.AudioManager.STREAM_ALARM */
#define SL_ANDROID_STREAM_ALARM        ((SLint32) 0x00000004)
/*      same as android.media.AudioManager.STREAM_NOTIFICATION */
#define SL_ANDROID_STREAM_NOTIFICATION ((SLint32) 0x00000005)
 
/*---------------------------------------------------------------------------*/
/* Android AudioPlayer and AudioRecorder configuration                       */
/*---------------------------------------------------------------------------*/
 
/** Audio Performance mode.
 * Performance mode tells the framework how to configure the audio path
 * for a player or recorder according to application performance and
 * functional requirements.
 * It affects the output or input latency based on acceptable tradeoffs on
 * battery drain and use of pre or post processing effects.
 * Performance mode should be set before realizing the object and should be
 * read after realizing the object to check if the requested mode could be
 * granted or not.
 */
/** Audio Performance mode key */
#define SL_ANDROID_KEY_PERFORMANCE_MODE ((const SLchar*) "androidPerformanceMode")
 
/** Audio performance values */
/*      No specific performance requirement. Allows HW and SW pre/post processing. */
#define SL_ANDROID_PERFORMANCE_NONE ((SLuint32) 0x00000000)
/*      Priority given to latency. No HW or software pre/post processing.
 *      This is the default if no performance mode is specified. */
#define SL_ANDROID_PERFORMANCE_LATENCY ((SLuint32) 0x00000001)
/*      Priority given to latency while still allowing HW pre and post processing. */
#define SL_ANDROID_PERFORMANCE_LATENCY_EFFECTS ((SLuint32) 0x00000002)
/*      Priority given to power saving if latency is not a concern.
 *      Allows HW and SW pre/post processing. */
#define SL_ANDROID_PERFORMANCE_POWER_SAVING ((SLuint32) 0x00000003)
在采集选项中包含

xxx_RECORDING_PRESET_GENERIC（通用配置，不知道是啥意思）
xxx_RECORDING_PRESET_CAMCORDER（录像中优先使用摄像头同方向的Mic，如果没有同方向的就使用主Mic）
xxx_RECORDING_PRESET_VOICE_RECOGNITION（针对语音识别业务进行了优化，可能使用降噪Mic）
xxx_RECORDING_PRESET_VOICE_COMMUNICATION（针对电话或网络电话优化，可能会硬件AEC、NS、AGC）
xxx_RECORDING_PRESET_UNPROCESSED（使用主Mic采集，不经过任何优化处理）

在渲染选项中包含

xxx_STREAM_VOICE（VoIP或者电话，音量需要通过通话音量调节）
xxx_STREAM_SYSTEM（系统音量，我的华为P10没有这个音量选项）
xxx_STREAM_RING（铃声音量）
xxx_STREAM_MEDIA（媒体音量）
xxx_STREAM_ALARM（闹钟音量）

————————————————
版权声明：本文为优快云博主「everlastxc」的原创文章，遵循CC 4.0 BY-SA版权协议，转载请附上原文出处链接及本声明。
原文链接：https://blog.youkuaiyun.com/qq_29621351/article/details/94562600
*/
SLAndroidConfigurationItf inputConfig;
result = (*recObjectItf_)
             ->GetInterface(recObjectItf_, SL_IID_ANDROIDCONFIGURATION,
                            &inputConfig);
if (SL_RESULT_SUCCESS == result) {
  SLuint32 presetValue = SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION;
  (*inputConfig)
      ->SetConfiguration(inputConfig, SL_ANDROID_KEY_RECORDING_PRESET,
                         &presetValue, sizeof(SLuint32));
}
result = (*recObjectItf_)->Realize(recObjectItf_, SL_BOOLEAN_FALSE);
SLASSERT(result);

//获取recordobjectItf

result =
    (*recObjectItf_)->GetInterface(recObjectItf_, SL_IID_RECORD, &recItf_);
SLASSERT(result);

//获取对象中的bufferqueue

result = (*recObjectItf_)
             ->GetInterface(recObjectItf_, SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
                            &recBufQueueItf_);
SLASSERT(result);

//注册bufferqueue中的callback回调接口

result = (*recBufQueueItf_)
             ->RegisterCallback(recBufQueueItf_, bqRecorderCallback, this);
SLASSERT(result);

//创建中转的缓冲区

devShadowQueue_ = new AudioQueue(DEVICE_SHADOW_BUFFER_QUEUE_LEN);
assert(devShadowQueue_);

AudioRecorder::Start

// in case already recording, stop recording and clear buffer queue，先暂停录制和情况buffer queue队列
result = (*recItf_)->SetRecordState(recItf_, SL_RECORDSTATE_STOPPED);
SLASSERT(result);
result = (*recBufQueueItf_)->Clear(recBufQueueItf_);
SLASSERT(result);

for (int i = 0; i < RECORD_DEVICE_KICKSTART_BUF_COUNT; i++) {
  sample_buf *buf = NULL;
  if (!freeQueue_->front(&buf)) {
    LOGE("=====OutOfFreeBuffers @ startingRecording @ (%d)", i);
    break;
  }
  freeQueue_->pop();
  assert(buf->buf_ && buf->cap_ && !buf->size_);

  result = (*recBufQueueItf_)->Enqueue(recBufQueueItf_, buf->buf_, buf->cap_);//把空闲的buffer，输入进去
  SLASSERT(result);
  devShadowQueue_->push(buf);//并把这个buffer放到这个中转队列
}

//设置录制状态为RECORDING

result = (*recItf_)->SetRecordState(recItf_, SL_RECORDSTATE_RECORDING);
SLASSERT(result);

AudioRecorder::ProcessSLCallback

assert(bq == recBufQueueItf_);
sample_buf *dataBuf = NULL;
devShadowQueue_->front(&dataBuf);
devShadowQueue_->pop();
dataBuf->size_ = dataBuf->cap_;  // device only calls us when it is really
                                 // full

callback_(ctx_, ENGINE_SERVICE_MSG_RECORDED_AUDIO_AVAILABLE, dataBuf);//callback，audio数据已经存在
recQueue_->push(dataBuf);

//重新设置空闲的buffer到如队列
sample_buf *freeBuf;
while (freeQueue_->front(&freeBuf) && devShadowQueue_->push(freeBuf)) {
  freeQueue_->pop();
  SLresult result = (*bq)->Enqueue(bq, freeBuf->buf_, freeBuf->cap_);
  SLASSERT(result);
}

Audio Delay

当信号输入进来时，使信号的输出波形比输入滞后设定的时间值比如100ms

把信号持续输入看成一个和时间相关的数据流，设定一个缓冲Buffer,大小根据delay的最大值设定，比如delay最大值为2000ms、那么对于96Khz采样频率的处理来说B u f f e r = 96 ∗ 2000 Buffer = 96 * 2000Buffer=96∗2000; Buffer的大小可能还需要加上DSP一次运算的样本点数比如128，这样最终的数据缓冲值B u f f e r = 192128 Buffer = 192128Buffer=192128;
设定一个结构体，内部含有缓冲区信号样本点输入计数器指针∗ i n P *inP∗inP和样本信号计数器输出指针∗ o u t P *outP∗outP、以及延迟设置值D l y DlyDly;
∗ o u t P *outP∗outP的值根据∗ i n P *inP∗inP和D l y DlyDly之差来确定，要注意∗ i n P *inP∗inP比D l y DlyDly小的情况，这种情况下 ∗ o u t P *outP∗outP的值应该取上一个Buffer的数据，即∗ o u t P *outP∗outP位于∗ i n P *inP∗inP上一个Buffer中而不是同属一个Buffer里面
————————————————
版权声明：本文为优快云博主「Flynn2019」的原创文章，遵循CC 4.0 BY-SA版权协议，转载请附上原文出处链接及本声明。
原文链接：https://blog.youkuaiyun.com/bentengdema/article/details/102495512

AudioDelay::AudioDelay

feedbackFactor_ = static_cast<int32_t>(decayWeight_ * kFloatToIntMapFactor);
liveAudioFactor_ = kFloatToIntMapFactor - feedbackFactor_;
allocateBuffer();

allocteBuffer

//转为s

float floatDelayTime = (float)delayTime_ / kMsPerSec;
//转为1s的总帧数
float fNumFrames = floatDelayTime * (float)sampleRate_ / kMsPerSec;
//总帧数
size_t sampleCount = static_cast<uint32_t>(fNumFrames + 0.5f) * channelCount_;

//字节数

uint32_t bytePerSample = format_ / 8;
assert(bytePerSample <= 4 && bytePerSample);

//每帧的字节数

uint32_t bytePerFrame = channelCount_ * bytePerSample;

// get bufCapacity in bytes
bufCapacity_ = sampleCount * bytePerSample;
bufCapacity_ =
    ((bufCapacity_ + bytePerFrame - 1) / bytePerFrame) * bytePerFrame;//按每帧的字节数对齐

buffer_ = new uint8_t[bufCapacity_];
assert(buffer_);

memset(buffer_, 0, bufCapacity_);
curPos_ = 0;

// bufSize_ is in Frames ( not samples, not bytes )
bufSize_ = bufCapacity_ / bytePerFrame;

AudioDelay::process

process() filter live audio with "echo" effect:
*   delay time is run-time adjustable
*   decay time could also be adjustable, but not used
*   in this sample, hardcoded to .5
*
* @param liveAudio is recorded audio stream
* @param channelCount for liveAudio, must be 2 for stereo
* @param numFrames is length of liveAudio in Frames ( not in byte )

// process every sample,处理每一帧
int32_t sampleCount = channelCount_ * numFrames;
int16_t* samples = &static_cast<int16_t*>(buffer_)[curPos_ * channelCount_];

for (size_t idx = 0; idx < sampleCount; idx++) {
#if 1
    int32_t curSample =
        (samples[idx] * feedbackFactor_ + liveAudio[idx] * liveAudioFactor_) /
        kFloatToIntMapFactor;

//当前帧和上一阵的根据相应的因素的叠加

    if (curSample > SHRT_MAX)
      curSample = SHRT_MAX;
    else if (curSample < SHRT_MIN)
      curSample = SHRT_MIN;
 
    liveAudio[idx] = samples[idx];
    samples[idx] = static_cast<int16_t>(curSample);
#else
    // Pure delay
    int16_t tmp = liveAudio[idx];
    liveAudio[idx] = samples[idx];
    samples[idx] = tmp;
#endif
  }

  curPos_ += numFrames;
  lock_.unlock();

Audio Common

void ConvertToSLSampleFormat(SLAndroidDataFormat_PCM_EX* pFormat,
                             SampleFormat* pSampleInfo_) {

// Only support 2 channels，设置声道数
// For channelMask, refer to wilhelm/src/android/channels.c for details
if (pSampleInfo_->channels_ <= 1) {
  pFormat->numChannels = 1;
  pFormat->channelMask = SL_SPEAKER_FRONT_LEFT;
} else {
  pFormat->numChannels = 2;
  pFormat->channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
}

case SL_ANDROID_PCM_REPRESENTATION_FLOAT:
  pFormat->bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_32;//设置每一帧的bit数
  pFormat->containerSize = SL_PCMSAMPLEFORMAT_FIXED_32;//设置container的大小
  pFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX;//设置数据类型
  break;

audio_main

分别创建audioplayer和audio record，同时注册了

engine.player_->SetBufQueue(engine.recBufQueue_, engine.freeBufQueue_);
engine.player_->RegisterCallback(EngineService, (void *)&engine);

engine.recorder_->SetBufQueues(engine.freeBufQueue_, engine.recBufQueue_);
engine.recorder_->RegisterCallback(EngineService, (void *)&engine);

bool EngineService(void *ctx, uint32_t msg, void *data) {
  assert(ctx == &engine);
  switch (msg) {
    case ENGINE_SERVICE_MSG_RETRIEVE_DUMP_BUFS: {
      *(static_cast<uint32_t *>(data)) = dbgEngineGetBufCount();
      break;
    }
    case ENGINE_SERVICE_MSG_RECORDED_AUDIO_AVAILABLE: {//当有录制的audio数据时
      // adding audio delay effect
      sample_buf *buf = static_cast<sample_buf *>(data);
      assert(engine.fastPathFramesPerBuf_ ==
             buf->size_ / engine.sampleChannels_ / (engine.bitsPerSample_ / 8));
      engine.delayEffect_->process(reinterpret_cast<int16_t *>(buf->buf_),
                                   engine.fastPathFramesPerBuf_);//达到延迟的效果
      break;
    }
    default:
      assert(false);
      return false;
  }

  return true;
}

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