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Java AudioFormat.ENCODING_PCM_8BIT屬性代碼示例

本文整理匯總了Java中android.media.AudioFormat.ENCODING_PCM_8BIT屬性的典型用法代碼示例。如果您正苦於以下問題：Java AudioFormat.ENCODING_PCM_8BIT屬性的具體用法？Java AudioFormat.ENCODING_PCM_8BIT怎麽用？Java AudioFormat.ENCODING_PCM_8BIT使用的例子？那麽, 這裏精選的屬性代碼示例或許可以為您提供幫助。您也可以進一步了解該屬性所在類android.media.AudioFormat的用法示例。


在下文中一共展示了AudioFormat.ENCODING_PCM_8BIT屬性的8個代碼示例，這些例子默認根據受歡迎程度排序。您可以為喜歡或者感覺有用的代碼點讚，您的評價將有助於係統推薦出更棒的Java代碼示例。

示例1: getMinBufferSize

private int getMinBufferSize(int sampleRate, int channelConfig, int audioFormat) {
    minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, audioFormat);
    // 解決異常IllegalArgumentException: Invalid audio buffer size
    int channelCount = 1;
    switch (channelConfig) {
        // AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
        case AudioFormat.CHANNEL_OUT_DEFAULT:
        case AudioFormat.CHANNEL_OUT_MONO:
        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
            channelCount = 1;
            break;
        case AudioFormat.CHANNEL_OUT_STEREO:
        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
            channelCount = 2;
            break;
        default:
            channelCount = Integer.bitCount(channelConfig);
    }
    // 判斷minBufferSize是否在範圍內，如果不在設定默認值為1152
    int frameSizeInBytes = channelCount * (audioFormat == AudioFormat.ENCODING_PCM_8BIT ? 1 : 2);
    if ((minBufferSize % frameSizeInBytes != 0) || (minBufferSize < 1)) {
        minBufferSize = 1152;
    }
    return minBufferSize;
}
開發者ID:dueros，項目名稱:dcs-sdk-java，代碼行數:25，代碼來源:AudioTrackPlayerImpl.java

示例2: audioFormatSampleBytes

public static int audioFormatSampleBytes(int f)
{
    switch (f)
    {
        case AudioFormat.ENCODING_PCM_8BIT:
            return 1;
        case AudioFormat.ENCODING_PCM_16BIT:
        case AudioFormat.ENCODING_DEFAULT:
            return 2;
        case AudioFormat.ENCODING_PCM_FLOAT:
            return 4;
        case AudioFormat.ENCODING_INVALID:
        default:
            return 0;
    }
}
開發者ID:hcmlab，項目名稱:ssj，代碼行數:16，代碼來源:Microphone.java

示例3: audioFormatSampleType

public static Cons.Type audioFormatSampleType(int f)
{
    switch (f)
    {
        case AudioFormat.ENCODING_PCM_8BIT:
            return Cons.Type.CHAR;
        case AudioFormat.ENCODING_PCM_16BIT:
        case AudioFormat.ENCODING_DEFAULT:
            return Cons.Type.SHORT;
        case AudioFormat.ENCODING_PCM_FLOAT:
            return Cons.Type.FLOAT;
        case AudioFormat.ENCODING_INVALID:
        default:
            return Cons.Type.UNDEF;
    }
}
開發者ID:hcmlab，項目名稱:ssj，代碼行數:16，代碼來源:Microphone.java

示例4: start

/**
 * 設置頻率
 * @param rate
 */
@SuppressWarnings("deprecation")
public void start(int rate){
	stop();
	if(rate>0){
		Hz=rate;
		waveLen = RATE / Hz;
		length = waveLen * Hz;
		audioTrack=new AudioTrack(AudioManager.STREAM_MUSIC, RATE,
				AudioFormat.CHANNEL_CONFIGURATION_STEREO, // CHANNEL_CONFIGURATION_MONO,
				AudioFormat.ENCODING_PCM_8BIT, length, AudioTrack.MODE_STREAM);
		//生成正弦波
		wave=SinWave.sin(wave, waveLen, length);
		if(audioTrack!=null){
			audioTrack.play();
		}
	}else{
		return;
	}
	
}
開發者ID:Becavalier，項目名稱:QRDataTransfer-Android，代碼行數:24，代碼來源:AudioTrackManager.java

示例5: 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);
}
開發者ID:Arjun-sna，項目名稱:Android-AudioRecorder-App，代碼行數:41，代碼來源:AudioSaveHelper.java

示例6: findAudioRecord

/**
 * 查找可用的音頻錄製器
 *
 * @return
 */
private AudioRecord findAudioRecord() {
    int[] samplingRates = new int[]{44100, 22050, 11025, 8000};
    int[] audioFormats = new int[]{
            AudioFormat.ENCODING_PCM_16BIT,
            AudioFormat.ENCODING_PCM_8BIT};
    int[] channelConfigs = new int[]{
            AudioFormat.CHANNEL_IN_STEREO,
            AudioFormat.CHANNEL_IN_MONO};

    for (int rate : samplingRates) {
        for (int format : audioFormats) {
            for (int config : channelConfigs) {
                try {
                    int bufferSize = AudioRecord.getMinBufferSize(rate, config, format);
                    if (bufferSize != AudioRecord.ERROR_BAD_VALUE) {
                        for (int source : AUDIO_SOURCES) {
                            AudioRecord recorder = new AudioRecord(source, rate, config, format, bufferSize * 4);
                            if (recorder.getState() == AudioRecord.STATE_INITIALIZED) {
                                return recorder;
                            }
                        }
                    }
                } catch (Exception e) {
                    Log.e(TAG, "Init AudioRecord Error." + Log.getStackTraceString(e));
                }
            }
        }
    }
    return null;
}
開發者ID:uestccokey，項目名稱:EZFilter，代碼行數:35，代碼來源:MediaAudioEncoder.java

示例7: bitsPerSample

@Override public byte bitsPerSample() {
  if (audioEncoding == AudioFormat.ENCODING_PCM_16BIT) {
    return 16;
  } else if (audioEncoding == AudioFormat.ENCODING_PCM_8BIT) {
    return 8;
  } else {
    return 16;
  }
}
開發者ID:kailash09dabhi，項目名稱:OmRecorder，代碼行數:9，代碼來源:AudioRecordConfig.java

示例8: start

public void start(HashMap<String, String> options, ModuleResultListener listener){
    int audioChannel = AudioFormat.CHANNEL_IN_STEREO;
    if (options.get("channel").equals("mono")) audioChannel = AudioFormat.CHANNEL_IN_MONO;
    int sampleRate = 22050;
    int audioBit = AudioFormat.ENCODING_PCM_16BIT;
    switch (options.get("quality")) {
        case "low":
            sampleRate = 8000;
            audioBit = AudioFormat.ENCODING_PCM_8BIT;
            break;
        case "high":
            sampleRate = 44100;
            audioBit = AudioFormat.ENCODING_PCM_16BIT;
            break;
    }

    if (mIsRecording) {
        if (mIsPausing) {
            if (mRecorder != null)mRecorder.resumeRecording();
            mIsPausing = false;
            listener.onResult(null);
        } else {
            listener.onResult(Util.getError(Constant.RECORDER_BUSY, Constant.RECORDER_BUSY_CODE));
        }
    } else {
        String time_str = new Date().getTime() + "";
        try {
            mFile = Util.getFile(time_str + ".wav");
        } catch (IOException e) {
            e.printStackTrace();
            listener.onResult(Util.getError(Constant.MEDIA_INTERNAL_ERROR, Constant.MEDIA_INTERNAL_ERROR_CODE));
        }
        mRecorder = OmRecorder.wav(
                new PullTransport.Default(getMic(audioBit, audioChannel, sampleRate), new PullTransport.OnAudioChunkPulledListener() {
                    @Override
                    public void onAudioChunkPulled(AudioChunk audioChunk) {

                    }
                }), mFile);
        mRecorder.startRecording();
        mIsRecording = true;
        listener.onResult(null);
    }
}
開發者ID:natjs，項目名稱:nat-recorder，代碼行數:44，代碼來源:RecorderModule.java


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