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Java AudioFormat.CHANNEL_IN_MONO属性代码示例

本文整理汇总了Java中android.media.AudioFormat.CHANNEL_IN_MONO属性的典型用法代码示例。如果您正苦于以下问题:Java AudioFormat.CHANNEL_IN_MONO属性的具体用法?Java AudioFormat.CHANNEL_IN_MONO怎么用?Java AudioFormat.CHANNEL_IN_MONO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android.media.AudioFormat的用法示例。


在下文中一共展示了AudioFormat.CHANNEL_IN_MONO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: updateBufferSize

void updateBufferSize(boolean pause) {
    synchronized (bufferSizeLock) {
        int samplesUpdate;

        if (pause) {
            // we need make buffer multiply of pitch.getPitchTime() (100 ms).
            // to prevent missing blocks from view otherwise:

            // file may contain not multiply 'samplesUpdate' count of samples. it is about 100ms.
            // we can't show on pitchView sorter then 100ms samples. we can't add partial sample because on
            // resumeRecording we have to apply rest of samplesUpdate or reload all samples again
            // from file. better then confusing user we cut them on next resumeRecording.

            long l = 1000;
            l = l / pitch.getPitchTime() * pitch.getPitchTime();
            samplesUpdate = (int) (l * sampleRate / 1000.0);
        } else {
            samplesUpdate = this.samplesUpdate;
        }

        bufferSize = RawSamples.CHANNEL_CONFIG == AudioFormat.CHANNEL_IN_MONO ? samplesUpdate : samplesUpdate * 2;
    }
}
 
开发者ID:NandagopalR,项目名称:Android-Audio-Recorder,代码行数:23,代码来源:RecordingActivity.java

示例2: 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;
    }
 
开发者ID:ThinkKeep,项目名称:EvilsLive,代码行数:25,代码来源:AudioCapture.java

示例3: generateTrack

public AudioTrack generateTrack(int sampleRate, short[] buf, int len) {
    int end = len;

    int c = 0;

    if (RawSamples.CHANNEL_CONFIG == AudioFormat.CHANNEL_IN_MONO)
        c = AudioFormat.CHANNEL_OUT_MONO;

    if (RawSamples.CHANNEL_CONFIG == AudioFormat.CHANNEL_IN_STEREO)
        c = AudioFormat.CHANNEL_OUT_STEREO;

    // old phones bug.
    // http://stackoverflow.com/questions/27602492
    //
    // with MODE_STATIC setNotificationMarkerPosition not called
    AudioTrack track = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
            c, RawSamples.AUDIO_FORMAT,
            len * (Short.SIZE / 8), AudioTrack.MODE_STREAM);
    track.write(buf, 0, len);
    if (track.setNotificationMarkerPosition(end) != AudioTrack.SUCCESS)
        throw new RuntimeException("unable to set marker");
    return track;
}
 
开发者ID:NandagopalR,项目名称:Android-Audio-Recorder,代码行数:23,代码来源:Sound.java

示例4: RawAudioRecorder

/**
 * <p>Instantiates a new recorder and sets the state to INITIALIZING.
 * In case of errors, no exception is thrown, but the state is set to ERROR.</p>
 *
 * <p>Android docs say: 44100Hz is currently the only rate that is guaranteed to work on all devices,
 * but other rates such as 22050, 16000, and 11025 may work on some devices.</p>
 *
 * @param audioSource Identifier of the audio source (e.g. microphone)
 * @param sampleRate Sample rate (e.g. 16000)
 */
public RawAudioRecorder(int audioSource, int sampleRate) {
	mSampleRate = sampleRate;
	// E.g. 1 second of 16kHz 16-bit mono audio takes 32000 bytes.
	mOneSec = RESOLUTION_IN_BYTES * CHANNELS * mSampleRate;
	mRecording = new byte[mOneSec * MAX_RECORDING_TIME_IN_SECS];
	try {
		setBufferSizeAndFramePeriod();


		mRecorder = new AudioRecord(audioSource, mSampleRate, AudioFormat.CHANNEL_IN_MONO, RESOLUTION, mBufferSize);
		if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN) {
			boolean agcAvailable = AutomaticGainControl.isAvailable();
			if(agcAvailable) {
                   AutomaticGainControl.create(mRecorder.getAudioSessionId());
               }
		}

		if (mRecorder.getState() != AudioRecord.STATE_INITIALIZED) {
			throw new Exception("AudioRecord initialization failed");
		}
		mBuffer = new byte[mFramePeriod * RESOLUTION_IN_BYTES * CHANNELS];
		setState(State.READY);
	} catch (Exception e) {
		release();
		setState(State.ERROR);
		if (e.getMessage() == null) {
			Log.e(LOG_TAG, "Unknown error occured while initializing recording");
		} else {
			Log.e(LOG_TAG, e.getMessage());
		}
	}
}
 
开发者ID:suelensilva,项目名称:KaldiAndroid,代码行数:42,代码来源:RawAudioRecorder.java

示例5: getAudioChannelConfig

/**
 * Get the normal supported audio channel setting
 *
 * @return Audio channels
 */
public static int getAudioChannelConfig() {
    switch (SpeechRecognizer.AUDIO_CHANNELS) {
        case 1:
            return AudioFormat.CHANNEL_IN_MONO;
    }

    return AudioFormat.CHANNEL_IN_MONO;
}
 
开发者ID:olami-developers,项目名称:olami-android-client-sdk,代码行数:13,代码来源:AudioRecordManager.java

示例6: 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];

}
 
开发者ID:lrannn,项目名称:SimpleRecorder,代码行数:37,代码来源:MainActivity.java

示例7: SaiyRecorder

/**
 * Constructor
 * <p>
 * Uses the most common application defaults
 */
public SaiyRecorder() {
    this.audioSource = MediaRecorder.AudioSource.VOICE_RECOGNITION;
    this.sampleRateInHz = 8000;
    this.channelConfig = AudioFormat.CHANNEL_IN_MONO;
    this.audioFormat = AudioFormat.ENCODING_PCM_16BIT;
    this.bufferSizeInBytes = calculateBufferSize();
    this.enhance = true;
}
 
开发者ID:brandall76,项目名称:Saiy-PS,代码行数:13,代码来源:SaiyRecorder.java

示例8: AudioRecordRunnable

private AudioRecordRunnable() {
    bufferSize = AudioRecord.getMinBufferSize(sampleRate,
            AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT);
    audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, sampleRate,
            AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize);
    audioData = new short[bufferSize];
}
 
开发者ID:feigxj,项目名称:VideoRecorder-master,代码行数:7,代码来源:FFmpegRecorderActivity.java

示例9: SpeechRecord

public SpeechRecord(int sampleRateInHz, int bufferSizeInBytes, boolean noise, boolean gain, boolean echo)
        throws IllegalArgumentException {

    this(
            MediaRecorder.AudioSource.VOICE_RECOGNITION,
            sampleRateInHz,
            AudioFormat.CHANNEL_IN_MONO,
            AudioFormat.ENCODING_PCM_16BIT,
            bufferSizeInBytes,
            noise,
            gain,
            echo
    );
}
 
开发者ID:vaibhavs4424,项目名称:AI-Powered-Intelligent-Banking-Platform,代码行数:14,代码来源:SpeechRecord.java

示例10: getMinInputFrameSize

private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
  final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
  final int channelConfig =
      (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
  return AudioRecord.getMinBufferSize(
             sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
      / bytesPerFrame;
}
 
开发者ID:Piasy,项目名称:AppRTC-Android,代码行数:8,代码来源:WebRtcAudioManager.java

示例11: startRecording

/**
 * The trigger to open a new AudioRecord and start recording with the intention of sending the audio to the AVS server using the stopRecord(). This will have permissions
 * issues in Marshmallow that need to be handled at the Activity level (checking for permission to record audio, and requesting it if we don't already have permissions).
 * @param url our POST url
 * @param accessToken our user's access token
 * @param buffer a byte[] that allows us to prepend whatever audio is recorded by the user with either generated ore pre-recorded audio, this needs
 *               to be in the same format as the audio being recorded
 * @param callback our callback to notify us when we change states
 * @throws IOException
 *
 * @deprecated Manage this state on the application side, instead, and send the audio using {@link SpeechSendAudio}
 */
@Deprecated
public void startRecording(final String url, final String accessToken, @Nullable byte[] buffer,
                           @Nullable final AsyncCallback<Void, Exception> callback) throws IOException {
    synchronized(mLock) {
        mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, AUDIO_RATE, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, BUFFER_SIZE);
    }

    if(callback != null){
        callback.start();
    }

    mCallback = callback;
    mIsRecording = true;
    new AsyncTask<Void, Void, Void>() {
        @Override
        protected Void doInBackground(Void... params) {
            synchronized(mLock) {
                prepareConnection(url, accessToken);
            }
            return null;
        }
    }.executeOnExecutor(AsyncTask.THREAD_POOL_EXECUTOR);

    if(buffer != null){
        mOutputStream.write(buffer);
    }

    //record our audio
    recordAudio(mAudioRecord, mOutputStream);
}
 
开发者ID:vaibhavs4424,项目名称:AI-Powered-Intelligent-Banking-Platform,代码行数:42,代码来源:SpeechSendVoice.java

示例12: SpeechRecorder

SpeechRecorder(Conversation conversation) {
    mConversation = conversation;
    // audio should be 16-bit mono linear PCM at 16 kHz sample rate
    mRecorder = new AudioRecord(
            MediaRecorder.AudioSource.MIC,
            SAMPLE_RATE,
            AudioFormat.CHANNEL_IN_MONO,
            AudioFormat.ENCODING_PCM_16BIT,
            BUFFER_SIZE * BYTES_PER_SAMPLE
    );
}
 
开发者ID:pullstring,项目名称:pullstring-android,代码行数:11,代码来源:SpeechRecorder.java

示例13: findAudioRecord

/**
 * 查找可用的音频录制器
 *
 * @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;
}
 
开发者ID:uestccokey,项目名称:EZFilter,代码行数:35,代码来源:MediaAudioEncoder.java

示例14: Looper

public Looper() {
  minBytes = AudioRecord.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_IN_MONO,
      AudioFormat.ENCODING_PCM_16BIT);
  minBytes = Math.max(minBytes, fftBins);
  // VOICE_RECOGNITION: use the mic with AGC turned off!
  record =  new AudioRecord(AGC_OFF, sampleRate,
      AudioFormat.CHANNEL_IN_MONO,AudioFormat.ENCODING_PCM_16BIT,  minBytes);
  Log.d(TAG, "Buffer size: " + minBytes + " (" + record.getSampleRate() + "=" + sampleRate + ")");
}
 
开发者ID:neurobin,项目名称:audio-analyzer-for-android,代码行数:9,代码来源:AnalyzeActivity.java

示例15: setupAudioRecord

private void setupAudioRecord() {
    int channelConfig = stereo ? AudioFormat.CHANNEL_IN_STEREO : AudioFormat.CHANNEL_IN_MONO;
    int minBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig,
            AudioFormat.ENCODING_PCM_16BIT);
    audioRecord = new AudioRecord(
            MediaRecorder.AudioSource.MIC,
            sampleRate,
            channelConfig,
            AudioFormat.ENCODING_PCM_16BIT,
            4 * minBufferSize);
}
 
开发者ID:iwo,项目名称:SCR-Screen-Recorder-app,代码行数:11,代码来源:ProjectionThread.java


注:本文中的android.media.AudioFormat.CHANNEL_IN_MONO属性示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台,相关代码片段筛选自各路编程大神贡献的开源项目,源码版权归原作者所有,传播和使用请参考对应项目的License;未经允许,请勿转载。