android.media.AudioRecord#ERROR_BAD_VALUE源码实例Demo

下面列出了android.media.AudioRecord#ERROR_BAD_VALUE 实例代码，或者点击链接到github查看源代码，也可以在右侧发表评论。

源代码1 项目： PLDroidRTCStreaming 文件： ExtAudioCapture.java

@Override
public void run() {
    while (!mIsLoopExit) {
        byte[] buffer = new byte[SAMPLES_PER_FRAME * 2];
        int ret = mAudioRecord.read(buffer, 0, buffer.length);
        if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
            Log.e(TAG, "Error ERROR_INVALID_OPERATION");
        } else if (ret == AudioRecord.ERROR_BAD_VALUE) {
            Log.e(TAG, "Error ERROR_BAD_VALUE");
        } else {
            if (mOnAudioFrameCapturedListener != null) {
                mOnAudioFrameCapturedListener.onAudioFrameCaptured(buffer, System.nanoTime());
            }
        }
    }
}

源代码2 项目： Android 文件： AudioCapturer.java

@Override
public void run() {

    while (!mIsLoopExit) {

        byte[] buffer = new byte[mMinBufferSize];

        int ret = mAudioRecord.read(buffer, 0, mMinBufferSize);				
        if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
            Log.e(TAG , "Error ERROR_INVALID_OPERATION");
        } 
        else if (ret == AudioRecord.ERROR_BAD_VALUE) {
            Log.e(TAG , "Error ERROR_BAD_VALUE");
        } 
        else { 
            if (mAudioFrameCapturedListener != null) {
                mAudioFrameCapturedListener.onAudioFrameCaptured(buffer);
            }   
            Log.d(TAG , "OK, Captured "+ret+" bytes !");
        }			
        
        SystemClock.sleep(10);
    }		
}

源代码3 项目： black-mirror 文件： VoiceRecorder.java

/**
 * Creates a new {@link AudioRecord}.
 *
 * @return A newly created {@link AudioRecord}, or null if it cannot be created (missing
 * permissions?).
 */
private AudioRecord createAudioRecord() {
    for (int sampleRate : SAMPLE_RATE_CANDIDATES) {
        final int sizeInBytes = AudioRecord.getMinBufferSize(sampleRate, CHANNEL, ENCODING);
        if (sizeInBytes == AudioRecord.ERROR_BAD_VALUE) {
            continue;
        }
        final AudioRecord audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC,
                sampleRate, CHANNEL, ENCODING, sizeInBytes);
        if (audioRecord.getState() == AudioRecord.STATE_INITIALIZED) {
            mBuffer = new byte[sizeInBytes];
            return audioRecord;
        } else {
            audioRecord.release();
        }
    }
    return null;
}

源代码4 项目： AudioVideoCodec 文件： AudioCapture.java

public void start(int audioSource, int sampleRate, int channels, int audioFormat) {
    if (isStartRecord) {
        if (isDebug)
            Log.d(TAG, "音频录制已经开启");
        return;
    }

    //各厂商实现存在差异
    bufferSize = AudioRecord.getMinBufferSize(sampleRate, channels, audioFormat);

    if (bufferSize == AudioRecord.ERROR_BAD_VALUE) {
        if (isDebug)
            Log.d(TAG, "无效参数");
        return;
    }

    if (isDebug)
        Log.d(TAG, "bufferSize = ".concat(String.valueOf(bufferSize)).concat("byte"));

    audioRecord = new AudioRecord(AudioCapture.AUDIO_SOURCE, sampleRate, channels, audioFormat, bufferSize);

    audioRecord.startRecording();

    isStopRecord = false;

    threadCapture = new Thread(new CaptureRunnable());
    threadCapture.start();

    isStartRecord = true;
    if (isDebug) {
        Log.d(TAG, "音频录制开启...");
    }
}

源代码5 项目： PLDroidMediaStreaming 文件： ExtAudioCapture.java

@Override
public void run() {
    while (!mIsLoopExit) {
        int ret = mAudioRecord.read(mAudioSrcBuffer, 0, mAudioSrcBuffer.length);
        if (ret == AudioRecord.ERROR_INVALID_OPERATION) {
            Log.e(TAG, "Error ERROR_INVALID_OPERATION");
        } else if (ret == AudioRecord.ERROR_BAD_VALUE) {
            Log.e(TAG, "Error ERROR_BAD_VALUE");
        } else {
            if (mOnAudioFrameCapturedListener != null) {
                mOnAudioFrameCapturedListener.onAudioFrameCaptured(mAudioSrcBuffer);
            }
        }
    }
}

源代码6 项目： RxAndroidAudio 文件： StreamAudioRecorder.java

private void onError(int errorCode) {
    if (errorCode == AudioRecord.ERROR_INVALID_OPERATION) {
        Log.w(TAG, "record fail: ERROR_INVALID_OPERATION");
        mAudioDataCallback.onError();
    } else if (errorCode == AudioRecord.ERROR_BAD_VALUE) {
        Log.w(TAG, "record fail: ERROR_BAD_VALUE");
        mAudioDataCallback.onError();
    }
}

源代码7 项目： PermissionAgent 文件： RecordAudioTester.java

private static AudioRecord findAudioRecord() {
    for (int rate : RATES) {
        for (short format : new short[] {AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT}) {
            for (short channel : new short[] {AudioFormat.CHANNEL_IN_MONO, AudioFormat.CHANNEL_IN_STEREO}) {
                int buffer = AudioRecord.getMinBufferSize(rate, channel, format);
                if (buffer != AudioRecord.ERROR_BAD_VALUE) {
                    AudioRecord recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, rate, channel, format,
                        buffer);
                    if (recorder.getState() == AudioRecord.STATE_INITIALIZED) return recorder;
                }
            }
        }
    }
    return null;
}

源代码8 项目： cineio-broadcast-android 文件： MicrophoneEncoder.java

private void sendAudioToEncoder(boolean endOfStream) {
        // send current frame data to encoder
        try {
            ByteBuffer[] inputBuffers = mMediaCodec.getInputBuffers();
            audioInputBufferIndex = mMediaCodec.dequeueInputBuffer(-1);
            if (audioInputBufferIndex >= 0) {
                ByteBuffer inputBuffer = inputBuffers[audioInputBufferIndex];
                inputBuffer.clear();
                audioInputLength = mAudioRecord.read(inputBuffer, SAMPLES_PER_FRAME * 2);
                audioAbsolutePtsUs = (System.nanoTime()) / 1000L;
                // We divide audioInputLength by 2 because audio samples are
                // 16bit.
                audioAbsolutePtsUs = getJitterFreePTS(audioAbsolutePtsUs, audioInputLength / 2);

                if (audioInputLength == AudioRecord.ERROR_INVALID_OPERATION)
                    Log.e(TAG, "Audio read error: invalid operation");
                if (audioInputLength == AudioRecord.ERROR_BAD_VALUE)
                    Log.e(TAG, "Audio read error: bad value");
//                if (VERBOSE)
//                    Log.i(TAG, "queueing " + audioInputLength + " audio bytes with pts " + audioAbsolutePtsUs);
                if (endOfStream) {
                    if (VERBOSE) Log.i(TAG, "EOS received in sendAudioToEncoder");
                    mMediaCodec.queueInputBuffer(audioInputBufferIndex, 0, audioInputLength, audioAbsolutePtsUs, MediaCodec.BUFFER_FLAG_END_OF_STREAM);
                } else {
                    mMediaCodec.queueInputBuffer(audioInputBufferIndex, 0, audioInputLength, audioAbsolutePtsUs, 0);
                }
            }
        } catch (Throwable t) {
            Log.e(TAG, "_offerAudioEncoder exception");
            t.printStackTrace();
        }
    }

源代码9 项目： CameraV 文件： AACHelper.java

private int initAudioRecord(int rate)
{
    try
    {
        Log.v("===========Attempting rate ", rate + "Hz, bits: " + audioFormat + ", channel: " + channelConfig);
        bufferSize = AudioRecord.getMinBufferSize(rate, channelConfig, audioFormat);

        if (bufferSize != AudioRecord.ERROR_BAD_VALUE)
        {
            // check if we can instantiate and have a success
            recorder = new AudioRecord(AudioSource.MIC, rate, channelConfig, audioFormat, bufferSize);

            if (recorder.getState() == AudioRecord.STATE_INITIALIZED)
            {
                Log.v("===========final rate ", rate + "Hz, bits: " + audioFormat + ", channel: " + channelConfig);

                return rate;
            }
        }
    }
    catch (Exception e)
    {
        Log.v("error", "" + rate);
    }

    return -1;
}

源代码10 项目： NoiseCapture 文件： AudioProcess.java

/**
 * Constructor
 * @param recording Recording state
 * @param canceled Canceled state
 * @param customLeqProcessing Custom receiver of sound signals
 */
public AudioProcess(AtomicBoolean recording, AtomicBoolean canceled, ProcessingThread
        customLeqProcessing) {
    this.recording = recording;
    this.canceled = canceled;
    this.customLeqProcessing = customLeqProcessing;
    final int[] mSampleRates = new int[] {44100}; // AWeigting coefficient are based on 44100
    // Hz sampling rate, so we do not support other samplings (22050, 16000, 11025,8000)
    final int[] encodings = new int[] { AudioFormat.ENCODING_PCM_16BIT , AudioFormat.ENCODING_PCM_8BIT };
    final short[] audioChannels = new short[] { AudioFormat.CHANNEL_IN_MONO, AudioFormat.CHANNEL_IN_STEREO };
    for (int tryRate : mSampleRates) {
        for (int tryEncoding : encodings) {
            for(short tryAudioChannel : audioChannels) {
                int tryBufferSize = AudioRecord.getMinBufferSize(tryRate,
                        tryAudioChannel, tryEncoding);
                if (tryBufferSize != AudioRecord.ERROR_BAD_VALUE) {
                    // Take a higher buffer size in order to get a smooth recording under load
                    // avoiding Buffer overflow error on AudioRecord side.
                    bufferSize = Math.max(tryBufferSize,
                            (int)(AcousticIndicators.TIMEPERIOD_FAST * tryRate));
                    encoding = tryEncoding;
                    audioChannel = tryAudioChannel;
                    rate = tryRate;
                    this.fastLeqProcessing = new LeqProcessingThread(this,
                            AcousticIndicators.TIMEPERIOD_FAST, true,
                            hannWindowFast ? FFTSignalProcessing.WINDOW_TYPE.TUKEY :
                                    FFTSignalProcessing.WINDOW_TYPE.RECTANGULAR, PROP_MOVING_SPECTRUM, true);
                    this.slowLeqProcessing = new LeqProcessingThread(this,
                            AcousticIndicators.TIMEPERIOD_SLOW, true,
                            hannWindowOneSecond ? FFTSignalProcessing.WINDOW_TYPE.TUKEY :
                                    FFTSignalProcessing.WINDOW_TYPE.RECTANGULAR,
                            PROP_DELAYED_STANDART_PROCESSING, false);
                    return;
                }
            }
        }
    }
    throw new IllegalStateException("This device is not compatible");
}

源代码11 项目： PLDroidRTCStreaming 文件： ExtAudioCapture.java

public boolean startCapture(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat) {
    if (mIsCaptureStarted) {
        Log.e(TAG, "Capture already started !");
        return false;
    }

    int minBufferSize = AudioRecord.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);
    if (minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
        Log.e(TAG, "Invalid parameter !");
        return false;
    }

    mAudioRecord = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat, minBufferSize * 4);
    if (mAudioRecord.getState() == AudioRecord.STATE_UNINITIALIZED) {
        Log.e(TAG, "AudioRecord initialize fail !");
        return false;
    }

    mAudioRecord.startRecording();

    mIsLoopExit = false;
    mCaptureThread = new Thread(new AudioCaptureRunnable());
    mCaptureThread.start();

    mIsCaptureStarted = true;

    Log.d(TAG, "Start audio capture success !");

    return true;
}

源代码12 项目： AndroidInstantVideo 文件： AACEncoder.java

public void start() {
    mAudioRecord.startRecording();
    mMediaCodec.start();
    final long startWhen = System.nanoTime();
    final ByteBuffer[] inputBuffers = mMediaCodec.getInputBuffers();
    mThread = new Thread(new Runnable() {
        @Override
        public void run() {
            int len, bufferIndex;
            while (isStart && !Thread.interrupted()) {
                synchronized (mMediaCodec) {
                    if (!isStart) return;
                    bufferIndex = mMediaCodec.dequeueInputBuffer(10000);
                    if (bufferIndex >= 0) {
                        inputBuffers[bufferIndex].clear();
                        long presentationTimeNs = System.nanoTime();
                        len = mAudioRecord.read(inputBuffers[bufferIndex], bufferSize);
                        presentationTimeNs -= (len / samplingRate ) / 1000000000;
                        Loggers.i(TAG, "Index: " + bufferIndex + " len: " + len + " buffer_capacity: " + inputBuffers[bufferIndex].capacity());
                        long presentationTimeUs = (presentationTimeNs - startWhen) / 1000;
                        if (len == AudioRecord.ERROR_INVALID_OPERATION || len == AudioRecord.ERROR_BAD_VALUE) {
                            Log.e(TAG, "An error occured with the AudioRecord API !");
                        } else {
                            mMediaCodec.queueInputBuffer(bufferIndex, 0, len, presentationTimeUs, 0);
                            if (onDataComingCallback != null) {
                                onDataComingCallback.onComing();
                            }
                        }
                    }
                }
            }
        }
    });

    mThread.start();
    isStart = true;
}

源代码13 项目： EZFilter 文件： MediaAudioEncoder.java

/**
 * 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) {
                                mSamplingRate = rate;
                                return recorder;
                            }
                        }
                    }
                } catch (Exception e) {
                    Log.e(TAG, "Init AudioRecord Error." + Log.getStackTraceString(e));
                }
            }
        }
    }
    return null;
}

源代码14 项目： voice-pitch-analyzer 文件： SampleRateCalculator.java

public static int getMaxSupportedSampleRate() {
/*
 * Valid Audio Sample rates
 *
 * @see <a
 * href="http://en.wikipedia.org/wiki/Sampling_%28signal_processing%29"
 * >Wikipedia</a>
 */
    final int validSampleRates[] = new int[]{
            47250, 44100, 44056, 37800, 32000, 22050, 16000, 11025, 4800, 8000,};
/*
 * Selecting default audio input source for recording since
 * AudioFormat.CHANNEL_CONFIGURATION_DEFAULT is deprecated and selecting
 * default encoding format.
 */
    for (int i = 0; i < validSampleRates.length; i++) {
        int result = AudioRecord.getMinBufferSize(validSampleRates[i],
                android.media.AudioFormat.CHANNEL_IN_MONO,
                android.media.AudioFormat.ENCODING_PCM_16BIT);
        if (result != AudioRecord.ERROR
                && result != AudioRecord.ERROR_BAD_VALUE && result > 0) {
            // return the mininum supported audio sample rate
            return validSampleRates[i];
        }
    }
    // If none of the sample rates are supported return -1 handle it in
    // calling method
    return -1;
}

源代码15 项目： Android 文件： AudioCapturer.java

public boolean startCapture(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat) {

        if (mIsCaptureStarted) {
            Log.e(TAG, "Capture already started !");
            return false;
        }
    
        mMinBufferSize = AudioRecord.getMinBufferSize(sampleRateInHz,channelConfig,audioFormat);
        if (mMinBufferSize == AudioRecord.ERROR_BAD_VALUE) {
            Log.e(TAG, "Invalid parameter !");
            return false;
        }
        Log.d(TAG , "getMinBufferSize = "+mMinBufferSize+" bytes !");
		
        mAudioRecord = new AudioRecord(audioSource,sampleRateInHz,channelConfig,audioFormat,mMinBufferSize);				
        if (mAudioRecord.getState() == AudioRecord.STATE_UNINITIALIZED) {
    	    Log.e(TAG, "AudioRecord initialize fail !");
    	    return false;
        }		

        mAudioRecord.startRecording();

        mIsLoopExit = false;
        mCaptureThread = new Thread(new AudioCaptureRunnable());
        mCaptureThread.start();

        mIsCaptureStarted = true;

        Log.d(TAG, "Start audio capture success !");

        return true;
    }

源代码16 项目： OmRecorder 文件： PullTransport.java

@Override void startPoolingAndWriting(AudioRecord audioRecord, int pullSizeInBytes,
    OutputStream outputStream) throws IOException {
  AudioChunk audioChunk = new AudioChunk.Bytes(new byte[pullSizeInBytes]);
  while (pullableSource.isEnableToBePulled()) {
    audioChunk.readCount(audioRecord.read(audioChunk.toBytes(), 0, pullSizeInBytes));
    if (AudioRecord.ERROR_INVALID_OPERATION != audioChunk.readCount()
        && AudioRecord.ERROR_BAD_VALUE != audioChunk.readCount()) {
      if (onAudioChunkPulledListener != null) {
        postPullEvent(audioChunk);
      }
      writeAction.execute(audioChunk, outputStream);
    }
  }
}

源代码17 项目： connectivity-samples 文件： AudioRecorder.java

@Override
protected boolean validSize(int size) {
  return size != AudioRecord.ERROR && size != AudioRecord.ERROR_BAD_VALUE;
}

源代码18 项目： DeviceConnect-Android 文件： MicOpusRecorder.java

/**
 * 音声をレコードして、MediaCodec に渡します.
 */
private void recordAudio() throws NativeInterfaceException {
    int samplingRate = mSamplingRate.getValue();
    int channels = mChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO;
    int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
    int bufferSize = AudioRecord.getMinBufferSize(samplingRate, channels, audioFormat) * 4;
    int oneFrameDataCount = mSamplingRate.getValue() / mFrameSize.getFps();

    mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.DEFAULT,
            samplingRate,
            channels,
            audioFormat,
            bufferSize);

    if (mAudioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
        if (mAudioRecordCallback != null) {
            mAudioRecordCallback.onEncoderError();
        }
        return;
    }

    if (mUseAEC && AcousticEchoCanceler.isAvailable()) {
        // ノイズキャンセラー
        mEchoCanceler = AcousticEchoCanceler.create(mAudioRecord.getAudioSessionId());
        if (mEchoCanceler != null) {
            int ret = mEchoCanceler.setEnabled(true);
            if (ret != AudioEffect.SUCCESS) {
                if (DEBUG) {
                    Log.w(TAG, "AcousticEchoCanceler is not supported.");
                }
            }
        }
    }

    OpusEncoder opusEncoder = null;

    try {
        opusEncoder = new OpusEncoder(mSamplingRate, mChannels, mFrameSize, mBitRate, mApplication);

        mAudioRecord.startRecording();

        short[] emptyBuffer = new short[oneFrameDataCount];
        short[] pcmBuffer = new short[oneFrameDataCount];
        byte[] opusFrameBuffer = opusEncoder.bufferAllocate();
        while (!mStopFlag) {
            int readSize = mAudioRecord.read(pcmBuffer, 0, oneFrameDataCount);
            if (readSize > 0) {
                int opusFrameBufferLength;
                if (isMute()) {
                    opusFrameBufferLength = opusEncoder.encode(emptyBuffer, readSize, opusFrameBuffer);
                } else {
                    opusFrameBufferLength = opusEncoder.encode(pcmBuffer, readSize, opusFrameBuffer);
                }

                if (opusFrameBufferLength > 0 && mAudioRecordCallback != null) {
                    mAudioRecordCallback.onPeriodicNotification(opusFrameBuffer, opusFrameBufferLength);
                }
            } else if (readSize == AudioRecord.ERROR_INVALID_OPERATION) {
                if (DEBUG) {
                    Log.e(TAG, "Invalid operation error.");
                }
                break;
            } else if (readSize == AudioRecord.ERROR_BAD_VALUE) {
                if (DEBUG) {
                    Log.e(TAG, "Bad value error.");
                }
                break;
            } else if (readSize == AudioRecord.ERROR) {
                if (DEBUG) {
                    Log.e(TAG, "Unknown error.");
                }
                break;
            }
        }
    } finally {
        if (mEchoCanceler != null) {
            mEchoCanceler.release();
            mEchoCanceler = null;
        }

        if (opusEncoder != null) {
            opusEncoder.release();
        }
    }
}

源代码19 项目： libstreaming 文件： AACStream.java

@Override
@SuppressLint({ "InlinedApi", "NewApi" })
protected void encodeWithMediaCodec() throws IOException {

	final int bufferSize = AudioRecord.getMinBufferSize(mQuality.samplingRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT)*2;

	((AACLATMPacketizer)mPacketizer).setSamplingRate(mQuality.samplingRate);

	mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, mQuality.samplingRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize);
	mMediaCodec = MediaCodec.createEncoderByType("audio/mp4a-latm");
	MediaFormat format = new MediaFormat();
	format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm");
	format.setInteger(MediaFormat.KEY_BIT_RATE, mQuality.bitRate);
	format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1);
	format.setInteger(MediaFormat.KEY_SAMPLE_RATE, mQuality.samplingRate);
	format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
	format.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, bufferSize);
	mMediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
	mAudioRecord.startRecording();
	mMediaCodec.start();

	final MediaCodecInputStream inputStream = new MediaCodecInputStream(mMediaCodec);
	final ByteBuffer[] inputBuffers = mMediaCodec.getInputBuffers();

	mThread = new Thread(new Runnable() {
		@Override
		public void run() {
			int len = 0, bufferIndex = 0;
			try {
				while (!Thread.interrupted()) {
					bufferIndex = mMediaCodec.dequeueInputBuffer(10000);
					if (bufferIndex>=0) {
						inputBuffers[bufferIndex].clear();
						len = mAudioRecord.read(inputBuffers[bufferIndex], bufferSize);
						if (len ==  AudioRecord.ERROR_INVALID_OPERATION || len == AudioRecord.ERROR_BAD_VALUE) {
							Log.e(TAG,"An error occured with the AudioRecord API !");
						} else {
							//Log.v(TAG,"Pushing raw audio to the decoder: len="+len+" bs: "+inputBuffers[bufferIndex].capacity());
							mMediaCodec.queueInputBuffer(bufferIndex, 0, len, System.nanoTime()/1000, 0);
						}
					}
				}
			} catch (RuntimeException e) {
				e.printStackTrace();
			}
		}
	});

	mThread.start();

	// The packetizer encapsulates this stream in an RTP stream and send it over the network
	mPacketizer.setInputStream(inputStream);
	mPacketizer.start();

	mStreaming = true;

}

源代码20 项目： EvilsLive 文件： AudioCapture.java

public boolean startCapture(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat) {

        if (mIsCaptureStarted) {
            Log.e(TAG, "hujd Capture already started !");
            return false;
        }

        mMinBufferSize = AudioRecord.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);

        if (mMinBufferSize == AudioRecord.ERROR_BAD_VALUE) {
            Log.e(TAG, "hujd Invalid parameter !");
            return false;
        }

        Log.e(TAG, "hujd getMinBufferSize = " + mMinBufferSize + " bytes !");

        mAudioRecord = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat, mMinBufferSize);

        if (mAudioRecord.getState() == AudioRecord.STATE_UNINITIALIZED) {
            Log.e(TAG, "hujd AudioRecord initialize fail !");
            return false;
        }

        mAudioRecord.startRecording();


        mIsLoopExit = false;
        mCaptureThread = new Thread(new AudioCaptureRunnable());

        mCaptureThread.start();

        mIsCaptureStarted = true;

        Log.e(TAG, "hujd Start audio capture success !");
        return true;

    }

方法所在类

android.media.AudioRecord