本文整理汇总了Java中android.media.AudioFormat.ENCODING_PCM_16BIT属性的典型用法代码示例。如果您正苦于以下问题:Java AudioFormat.ENCODING_PCM_16BIT属性的具体用法?Java AudioFormat.ENCODING_PCM_16BIT怎么用?Java AudioFormat.ENCODING_PCM_16BIT使用的例子?那么, 这里精选的属性代码示例或许可以为您提供帮助。您也可以进一步了解该属性所在类android.media.AudioFormat
的用法示例。
在下文中一共展示了AudioFormat.ENCODING_PCM_16BIT属性的15个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的Java代码示例。
示例1: getMinBufferSize
private int getMinBufferSize(int sampleRate, int channelConfig, int audioFormat) {
int numOfChannels, bitsPersample;
if (channelConfig == AudioFormat.CHANNEL_IN_MONO) {
numOfChannels = 1;
} else {
numOfChannels = 2;
}
if (AudioFormat.ENCODING_PCM_16BIT == audioFormat) {
bitsPersample = 16;
} else {
bitsPersample = 8;
}
int periodInFrames = sampleRate * TIMER_INTERVAL / 1000; //num of frames in a second is same as sample rate
//refer to android/4.1.1/frameworks/av/media/libmedia/AudioRecord.cpp, AudioRecord::getMinFrameCount method
//we times 2 for ping pong use of record buffer
mMinBufferSize = periodInFrames * 2 * numOfChannels * bitsPersample / 8;
if (mMinBufferSize < AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat)) {
// Check to make sure buffer size is not smaller than the smallest allowed one
mMinBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat);
// Set frame period and timer interval accordingly
// periodInFrames = mMinBufferSize / ( 2 * bitsPersample * numOfChannels / 8 );
}
return mMinBufferSize;
}
示例2: playSound
/**
* This method plays the sound data in the specified buffer.
*
* @param buffer specifies the sound data buffer.
*/
public void playSound(short[] buffer)
{
final String funcName = "playSound";
if (debugEnabled)
{
dbgTrace.traceEnter(funcName, TrcDbgTrace.TraceLevel.API);
dbgTrace.traceExit(funcName, TrcDbgTrace.TraceLevel.API);
}
audioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
sampleRate,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
buffer.length*2, //buffer length in bytes
AudioTrack.MODE_STATIC);
audioTrack.write(buffer, 0, buffer.length);
audioTrack.setNotificationMarkerPosition(buffer.length);
audioTrack.setPlaybackPositionUpdateListener(this);
audioTrack.play();
playing = true;
}
示例3: AudioSink
/**
* Constructor. Will create a new AudioSink.
*
* @param packetSize size of the incoming packets
* @param sampleRate sample rate of the audio signal
*/
public AudioSink (int packetSize, int sampleRate) {
this.packetSize = packetSize;
this.sampleRate = sampleRate;
// Create the queues and fill them with
this.inputQueue = new ArrayBlockingQueue<SamplePacket>(QUEUE_SIZE);
this.outputQueue = new ArrayBlockingQueue<SamplePacket>(QUEUE_SIZE);
for (int i = 0; i < QUEUE_SIZE; i++)
this.outputQueue.offer(new SamplePacket(packetSize));
// Create an instance of the AudioTrack class:
int bufferSize = AudioTrack.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
this.audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
// Create the audio filters:
this.audioFilter1 = FirFilter.createLowPass(2, 1, 1, 0.1f, 0.15f, 30);
Log.d(LOGTAG,"constructor: created audio filter 1 with " + audioFilter1.getNumberOfTaps() + " Taps.");
this.audioFilter2 = FirFilter.createLowPass(4, 1, 1, 0.1f, 0.1f, 30);
Log.d(LOGTAG,"constructor: created audio filter 2 with " + audioFilter2.getNumberOfTaps() + " Taps.");
this.tmpAudioSamples = new SamplePacket(packetSize);
}
示例4: PcmPlayer
public PcmPlayer(Context context, Handler handler) {
this.mContext = context;
this.audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, wBufferSize, AudioTrack.MODE_STREAM);
this.handler = handler;
audioTrack.setPlaybackPositionUpdateListener(this, handler);
cacheDir = context.getExternalFilesDir(Environment.DIRECTORY_MUSIC);
}
示例5: findAudioRecord
public AudioRecord findAudioRecord() {
for (int rate : mSampleRates) {
for (short audioFormat : new short[] { AudioFormat.ENCODING_PCM_16BIT }) {
for (short channelConfig : new short[] { AudioFormat.CHANNEL_IN_MONO }) {
try {
Log.d("C.TAG", "Attempting rate " + rate + "Hz, bits: " + audioFormat + ", channel: "
+ channelConfig);
int bufferSize = AudioRecord.getMinBufferSize(rate, AudioFormat.CHANNEL_IN_MONO , AudioFormat.ENCODING_PCM_16BIT);
if (bufferSize != AudioRecord.ERROR_BAD_VALUE) {
// check if we can instantiate and have a success
AudioRecord recorder = new AudioRecord(AudioSource.MIC, DEFAULT_RATE, channelConfig, audioFormat, bufferSize);
if (recorder.getState() == AudioRecord.STATE_INITIALIZED)
return recorder;
}
} catch (Exception e) {
Log.e("C.TAG", rate + "Exception, keep trying.",e);
}
}
}
}
return null;
}
示例6: AudioRecorder
public AudioRecorder(LoudnessSensor sensor) {
this.mSensor = sensor;
int channel = AudioFormat.CHANNEL_IN_MONO;
int mic = AudioSource.MIC;
// Berechne den Puffer
int minAudioBuffer = AudioRecord.getMinBufferSize(
COMMON_AUDIO_FREQUENCY,
channel,
AudioFormat.ENCODING_PCM_16BIT);
int audioBuffer = minAudioBuffer * 6;
// Erstelle den Recorder
audioInput = new AudioRecord(
mic,
COMMON_AUDIO_FREQUENCY,
channel,
AudioFormat.ENCODING_PCM_16BIT,
audioBuffer);
}
示例7: getInstance
public static ExtAudioRecorder getInstance(Boolean recordingCompressed, VoiceCallback callback) {
if (recordingCompressed) {
result = new ExtAudioRecorder(false, AudioSource.MIC,
sampleRates[3], AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, callback);
} else {
int i = 3;
do {
result = new ExtAudioRecorder(true, AudioSource.MIC,
sampleRates[i], AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, callback);
} while ((--i >= 0)
&& !(result.getState() == ExtAudioRecorder.State.INITIALIZING));
}
return result;
}
示例8: start
public void start() {
int minBufferSize = AudioRecord.getMinBufferSize(mSampleRate, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT);
int targetSize = mSampleRate * mChannels; // 1 seconds buffer size
if (targetSize < minBufferSize) {
targetSize = minBufferSize;
}
if (audioCapture == null) {
try {
audioCapture = new AudioRecord(MediaRecorder.AudioSource.MIC,
mSampleRate,
AudioFormat.CHANNEL_IN_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
targetSize);
} catch (IllegalArgumentException e) {
audioCapture = null;
}
}
LiveJniLib.native_audio_init(mSampleRate, mChannels);
if ( audioCapture != null) {
audioCapture.startRecording();
AudioEncoder audioEncoder = new AudioEncoder();
audioEncoder.start();
}
}
示例9: createAudioTrack
private AudioTrack createAudioTrack(int sampleRate, int channelCount) {
int channelConfig = channelCount == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO;
int bufferSize = ((sampleRate * 2) * channelCount / 100) * 8;//最多缓冲80毫秒的数据
return new AudioTrack(
AudioManager.STREAM_MUSIC,
sampleRate,
channelConfig,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize,
AudioTrack.MODE_STREAM);
}
示例10: writeWavHeader
/**
* Writes the proper 44-byte RIFF/WAVE header to/for the given stream
* Two size fields are left empty/null since we do not yet know the final stream size
*
* @param out The stream to write the header to
* @param channelMask An AudioFormat.CHANNEL_* mask
* @param sampleRate The sample rate in hertz
* @param encoding An AudioFormat.ENCODING_PCM_* value
* @throws IOException
*/
private void writeWavHeader(OutputStream out, int channelMask, int sampleRate, int encoding)
throws IOException {
short channels;
switch (channelMask) {
case AudioFormat.CHANNEL_IN_MONO:
channels = 1;
break;
case AudioFormat.CHANNEL_IN_STEREO:
channels = 2;
break;
default:
throw new IllegalArgumentException("Unacceptable channel mask");
}
short bitDepth;
switch (encoding) {
case AudioFormat.ENCODING_PCM_8BIT:
bitDepth = 8;
break;
case AudioFormat.ENCODING_PCM_16BIT:
bitDepth = 16;
break;
case AudioFormat.ENCODING_PCM_FLOAT:
bitDepth = 32;
break;
default:
throw new IllegalArgumentException("Unacceptable encoding");
}
writeWavHeader(out, channels, sampleRate, bitDepth);
}
示例11: RecordAudioinBytes
public RecordAudioinBytes() throws IllegalArgumentException {
super(RecorderConstants.AudioSource, RecorderConstants.mSampleRate,AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT,RecorderConstants.bufferSize);
mOneSec = RESOLUTION_IN_BYTES*CHANNELS*RecorderConstants.mSampleRate;
mRecording = new byte[mOneSec*35];
// int bufferSize = getBufferSize();
// int framePeriod = bufferSize / (2 * RESOLUTION_IN_BYTES * CHANNELS);
// createRecorder(audioSource, sampleRateInHz, bufferSize);
checkthingsforrecoder();
mBuffer = new byte[RecorderConstants.framePeriod*RESOLUTION_IN_BYTES*CHANNELS];
}
示例12: createAudioTrack
private void createAudioTrack() throws InitializationException {
// The AudioTrack configurations parameters used here, are guaranteed to
// be supported on all devices.
// AudioFormat.CHANNEL_OUT_MONO should be used in place of deprecated
// AudioFormat.CHANNEL_CONFIGURATION_MONO, but it is not available for
// API level 3.
// Output buffer for playing should be as short as possible, so
// AudioBufferPlayed events are not invoked long before audio buffer is
// actually played. Also, when AudioTrack is stopped, it is filled with
// silence of length audioTrackBufferSizeInBytes. If the silence is too
// long, it causes a delay before the next recorded data starts playing.
audioTrackBufferSizeInBytes = AudioTrack.getMinBufferSize(
SpeechTrainerConfig.SAMPLE_RATE_HZ,
AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT);
if (audioTrackBufferSizeInBytes <= 0) {
throw new InitializationException("Failed to initialize playback.");
}
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
SpeechTrainerConfig.SAMPLE_RATE_HZ,
AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
audioTrackBufferSizeInBytes,
AudioTrack.MODE_STREAM);
if (audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
audioTrack = null;
throw new InitializationException("Failed to initialize playback.");
}
}
示例13: audioTrackInit
public int audioTrackInit() {
// Log.e(" ffff mediaplayer audiotrackinit start . sampleRateInHz:=" + sampleRateInHz + " channels:=" + channels );
audioTrackRelease();
int channelConfig = channels >= 2 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO;
try {
mAudioTrackBufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT, mAudioTrackBufferSize, AudioTrack.MODE_STREAM);
} catch (Exception e) {
mAudioTrackBufferSize = 0;
Log.e("audioTrackInit", e);
}
return mAudioTrackBufferSize;
}
示例14: SpeechRecord
public SpeechRecord(int sampleRateInHz, int bufferSizeInBytes)
throws IllegalArgumentException {
this(
MediaRecorder.AudioSource.VOICE_RECOGNITION,
sampleRateInHz,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSizeInBytes,
false,
false,
false
);
}
示例15: onCreate
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
mImageButton = (ImageButton) findViewById(R.id.action_image);
mImageButton.setOnClickListener(new View.OnClickListener() {
@Override
public void onClick(View v) {
if (mRecorder == null) {
return;
}
boolean recording = mRecorder.isRecording();
if (recording) {
((ImageButton) v).setImageResource(R.drawable.record);
mRecorder.stop();
} else {
((ImageButton) v).setImageResource(R.drawable.pause);
mRecorder.startRecording();
}
}
});
boolean result = createOutputFile();
if (!result) {
Toast.makeText(this, "创建文件失败~", Toast.LENGTH_SHORT).show();
}
mRecorder = new Recorder(44100,
AudioFormat.CHANNEL_IN_MONO/*单双声道*/,
AudioFormat.ENCODING_PCM_16BIT/*格式*/,
MediaRecorder.AudioSource.MIC/*AudioSource*/,
NUM_SAMPLES/*period*/,
this/*onDataChangeListener*/);
output = new byte[NUM_SAMPLES * 2];
}