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Java MediaCodec.BufferInfo方法代码示例

本文整理汇总了Java中android.media.MediaCodec.BufferInfo方法的典型用法代码示例。如果您正苦于以下问题:Java MediaCodec.BufferInfo方法的具体用法?Java MediaCodec.BufferInfo怎么用?Java MediaCodec.BufferInfo使用的例子?那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在android.media.MediaCodec的用法示例。


在下文中一共展示了MediaCodec.BufferInfo方法的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: handleCodecOutput

import android.media.MediaCodec; //导入方法依赖的package包/类
private void handleCodecOutput(MediaCodec mediaCodec,
                               ByteBuffer[] codecOutputBuffers,
                               MediaCodec.BufferInfo bufferInfo,
                               OutputStream outputStream)
    throws IOException
{
  int codecOutputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);

  while (codecOutputBufferIndex != MediaCodec.INFO_TRY_AGAIN_LATER) {
    if (codecOutputBufferIndex >= 0) {
      ByteBuffer encoderOutputBuffer = codecOutputBuffers[codecOutputBufferIndex];

      encoderOutputBuffer.position(bufferInfo.offset);
      encoderOutputBuffer.limit(bufferInfo.offset + bufferInfo.size);

      if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != MediaCodec.BUFFER_FLAG_CODEC_CONFIG) {
        byte[] header = createAdtsHeader(bufferInfo.size - bufferInfo.offset);


        outputStream.write(header);

        byte[] data = new byte[encoderOutputBuffer.remaining()];
        encoderOutputBuffer.get(data);
        outputStream.write(data);
      }

      encoderOutputBuffer.clear();

      mediaCodec.releaseOutputBuffer(codecOutputBufferIndex, false);
    }  else if (codecOutputBufferIndex== MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
      codecOutputBuffers = mediaCodec.getOutputBuffers();
    }

    codecOutputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);
  }

}
 
开发者ID:XecureIT,项目名称:PeSanKita-android,代码行数:38,代码来源:AudioCodec.java

示例2: writeFrameByte

import android.media.MediaCodec; //导入方法依赖的package包/类
private void writeFrameByte(int track, ByteBuffer bb, MediaCodec.BufferInfo bi, boolean isKeyFrame) {
    SrsEsFrame frame = new SrsEsFrame();
    frame.bb = bb;
    frame.bi = bi;
    frame.isKeyFrame = isKeyFrame;
    frame.track = track;

    if (bRecording && !bPaused) {
        if (needToFindKeyFrame) {
            if (frame.isKeyFrame) {
                needToFindKeyFrame = false;
                frameCache.add(frame);
                synchronized (writeLock) {
                    writeLock.notifyAll();
                }
            }
        } else {
            frameCache.add(frame);
            synchronized (writeLock) {
                writeLock.notifyAll();
            }
        }
    }
}
 
开发者ID:lisnstatic,项目名称:live_master,代码行数:25,代码来源:SrsMp4Muxer.java

示例3: srs_avc_startswith_annexb

import android.media.MediaCodec; //导入方法依赖的package包/类
public SrsAnnexbSearch srs_avc_startswith_annexb(ByteBuffer bb, MediaCodec.BufferInfo bi) {
    SrsAnnexbSearch as = new SrsAnnexbSearch();
    as.match = false;

    int pos = bb.position();
    while (pos < bi.size - 3) {
        // not match.
        if (bb.get(pos) != 0x00 || bb.get(pos + 1) != 0x00) {
            break;
        }

        // match N[00] 00 00 01, where N>=0
        if (bb.get(pos + 2) == 0x01) {
            as.match = true;
            as.nb_start_code = pos + 3 - bb.position();
            break;
        }

        pos++;
    }

    return as;
}
 
开发者ID:lisnstatic,项目名称:live_master,代码行数:24,代码来源:SrsMp4Muxer.java

示例4: CameraRecorder

import android.media.MediaCodec; //导入方法依赖的package包/类
public CameraRecorder(){
        mShowEGLHelper=new EglHelper();
//        mEncodeEGLHelper=new EGLHelper();
        mSem=new Semaphore(0);
        mAudioEncodeBufferInfo=new MediaCodec.BufferInfo();
        mVideoEncodeBufferInfo=new MediaCodec.BufferInfo();
    }
 
开发者ID:aiyaapp,项目名称:AAVT,代码行数:8,代码来源:CameraRecorder.java

示例5: writeSampleData

import android.media.MediaCodec; //导入方法依赖的package包/类
public void writeSampleData(SampleType sampleType, ByteBuffer byteBuf, MediaCodec.BufferInfo bufferInfo) {
    if (mStarted) {
        mMuxer.writeSampleData(getTrackIndexForSampleType(sampleType), byteBuf, bufferInfo);
        return;
    }
    byteBuf.limit(bufferInfo.offset + bufferInfo.size);
    byteBuf.position(bufferInfo.offset);
    if (mByteBuffer == null) {
        mByteBuffer = ByteBuffer.allocateDirect(BUFFER_SIZE).order(ByteOrder.nativeOrder());
    }
    mByteBuffer.put(byteBuf);
    mSampleInfoList.add(new SampleInfo(sampleType, bufferInfo.size, bufferInfo));
}
 
开发者ID:SavorGit,项目名称:Hotspot-master-devp,代码行数:14,代码来源:QueuedMuxer.java

示例6: writeMediaData

import android.media.MediaCodec; //导入方法依赖的package包/类
public synchronized void writeMediaData(int trackIndex, ByteBuffer byteBuf, MediaCodec.BufferInfo bufferInfo) {
    if (mMuxerStarted) {
        RecordScreenLogUtil.e(TAG, "当前线程::::" + trackIndex);
        if (mMuxer != null && mMuxerStarted == true)
            mMuxer.writeSampleData(trackIndex, byteBuf, bufferInfo);
    }
}
 
开发者ID:sunshinecoast,项目名称:ScreenRecordCaptureMaster,代码行数:8,代码来源:YXMuxerWrapper.java

示例7: EncoderCore

import android.media.MediaCodec; //导入方法依赖的package包/类
public EncoderCore(MMediaMuxer MMediaMuxer) {
    mMuxerStarted = false;
    mBufferInfo = new MediaCodec.BufferInfo();
    mTrackIndex = -1;
    this.MMediaMuxer = MMediaMuxer;
    this.mMuxer= MMediaMuxer.getMuxer();
}
 
开发者ID:zhangyaqiang,项目名称:Fatigue-Detection,代码行数:8,代码来源:EncoderCore.java

示例8: prepare

import android.media.MediaCodec; //导入方法依赖的package包/类
public void prepare(YXMuxerWrapper muxerWrapper) throws IOException {
    mMuxer = muxerWrapper;
    mBufferInfo = new MediaCodec.BufferInfo();

    mAudioEncoder = MediaCodec.createEncoderByType(MIME_TYPE);
    mAudioFormat = MediaFormat.createAudioFormat(MIME_TYPE, SAMPLE_RATE, CHANNEL);
    mAudioFormat.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE);
    mAudioFormat.setInteger(MediaFormat.KEY_CHANNEL_MASK, AudioFormat.CHANNEL_IN_STEREO);
    mAudioFormat.setInteger(MediaFormat.KEY_BIT_RATE, BIT_RATE);
    mAudioFormat.setInteger(MediaFormat.KEY_CHANNEL_COUNT, CHANNEL);
    mAudioFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, 8192);
    mAudioEncoder.configure(mAudioFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
}
 
开发者ID:sunshinecoast,项目名称:ScreenRecordCaptureMaster,代码行数:14,代码来源:YXAudioEncoder.java

示例9: writeToBufferInfo

import android.media.MediaCodec; //导入方法依赖的package包/类
private void writeToBufferInfo(MediaCodec.BufferInfo bufferInfo, int offset) {
    bufferInfo.set(offset, mSize, mPresentationTimeUs, mFlags);
}
 
开发者ID:SavorGit,项目名称:Hotspot-master-devp,代码行数:4,代码来源:QueuedMuxer.java

示例10: getAacDataRtp

import android.media.MediaCodec; //导入方法依赖的package包/类
@Override
protected void getAacDataRtp(ByteBuffer aacBuffer, MediaCodec.BufferInfo info) {
  rtspClient.sendAudio(aacBuffer, info);
}
 
开发者ID:pedroSG94,项目名称:rtmp-rtsp-stream-client-java,代码行数:5,代码来源:RtspCamera1.java

示例11: SampleInfo

import android.media.MediaCodec; //导入方法依赖的package包/类
private SampleInfo(SampleType sampleType, int size, MediaCodec.BufferInfo bufferInfo) {
    mSampleType = sampleType;
    mSize = size;
    mPresentationTimeUs = bufferInfo.presentationTimeUs;
    mFlags = bufferInfo.flags;
}
 
开发者ID:SavorGit,项目名称:Hotspot-master-devp,代码行数:7,代码来源:QueuedMuxer.java

示例12: onAudioEncode

import android.media.MediaCodec; //导入方法依赖的package包/类
@Override
public void onAudioEncode(ByteBuffer bb, MediaCodec.BufferInfo bi) {
    if (mPacker != null) {
        mPacker.onAudioData(bb, bi);
    }
}
 
开发者ID:wuyisheng,项目名称:libRtmp,代码行数:7,代码来源:DataWatcher.java

示例13: getH264DataRtp

import android.media.MediaCodec; //导入方法依赖的package包/类
@Override
protected void getH264DataRtp(ByteBuffer h264Buffer, MediaCodec.BufferInfo info) {
  srsFlvMuxer.sendVideo(h264Buffer, info);
}
 
开发者ID:pedroSG94,项目名称:rtmp-rtsp-stream-client-java,代码行数:5,代码来源:RtmpCamera1.java

示例14: dequeueOutputBuffer

import android.media.MediaCodec; //导入方法依赖的package包/类
private DecodedOutputBuffer dequeueOutputBuffer(int dequeueTimeoutMs) {
  checkOnMediaCodecThread();
  if (decodeStartTimeMs.isEmpty()) {
    return null;
  }
  // Drain the decoder until receiving a decoded buffer or hitting
  // MediaCodec.INFO_TRY_AGAIN_LATER.
  final MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
  while (true) {
    final int result =
        mediaCodec.dequeueOutputBuffer(info, TimeUnit.MILLISECONDS.toMicros(dequeueTimeoutMs));
    switch (result) {
      case MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED:
        outputBuffers = mediaCodec.getOutputBuffers();
        Logging.d(TAG, "Decoder output buffers changed: " + outputBuffers.length);
        if (hasDecodedFirstFrame) {
          throw new RuntimeException("Unexpected output buffer change event.");
        }
        break;
      case MediaCodec.INFO_OUTPUT_FORMAT_CHANGED:
        MediaFormat format = mediaCodec.getOutputFormat();
        Logging.d(TAG, "Decoder format changed: " + format.toString());
        int new_width = format.getInteger(MediaFormat.KEY_WIDTH);
        int new_height = format.getInteger(MediaFormat.KEY_HEIGHT);
        if (hasDecodedFirstFrame && (new_width != width || new_height != height)) {
          throw new RuntimeException("Unexpected size change. Configured " + width + "*" + height
              + ". New " + new_width + "*" + new_height);
        }
        width = format.getInteger(MediaFormat.KEY_WIDTH);
        height = format.getInteger(MediaFormat.KEY_HEIGHT);

        if (!useSurface && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
          colorFormat = format.getInteger(MediaFormat.KEY_COLOR_FORMAT);
          Logging.d(TAG, "Color: 0x" + Integer.toHexString(colorFormat));
          if (!supportedColorList.contains(colorFormat)) {
            throw new IllegalStateException("Non supported color format: " + colorFormat);
          }
        }
        if (format.containsKey("stride")) {
          stride = format.getInteger("stride");
        }
        if (format.containsKey("slice-height")) {
          sliceHeight = format.getInteger("slice-height");
        }
        Logging.d(TAG, "Frame stride and slice height: " + stride + " x " + sliceHeight);
        stride = Math.max(width, stride);
        sliceHeight = Math.max(height, sliceHeight);
        break;
      case MediaCodec.INFO_TRY_AGAIN_LATER:
        return null;
      default:
        hasDecodedFirstFrame = true;
        TimeStamps timeStamps = decodeStartTimeMs.remove();
        long decodeTimeMs = SystemClock.elapsedRealtime() - timeStamps.decodeStartTimeMs;
        if (decodeTimeMs > MAX_DECODE_TIME_MS) {
          Logging.e(TAG, "Very high decode time: " + decodeTimeMs + "ms"
                  + ". Q size: " + decodeStartTimeMs.size()
                  + ". Might be caused by resuming H264 decoding after a pause.");
          decodeTimeMs = MAX_DECODE_TIME_MS;
        }
        return new DecodedOutputBuffer(result, info.offset, info.size,
            TimeUnit.MICROSECONDS.toMillis(info.presentationTimeUs), timeStamps.timeStampMs,
            timeStamps.ntpTimeStampMs, decodeTimeMs, SystemClock.elapsedRealtime());
    }
  }
}
 
开发者ID:lgyjg,项目名称:AndroidRTC,代码行数:67,代码来源:MediaCodecVideoDecoder.java

示例15: writeAudioSample

import android.media.MediaCodec; //导入方法依赖的package包/类
public void writeAudioSample(final ByteBuffer bb, MediaCodec.BufferInfo bi) {
  int dts = (int) (bi.presentationTimeUs / 1000);

  audio_tag = mAudioAllocator.allocate(bi.size + 2);
  byte aac_packet_type = 1; // 1 = AAC raw
  if (!aac_specific_config_got) {
    // @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf
    // AudioSpecificConfig (), page 33
    // 1.6.2.1 AudioSpecificConfig
    // audioObjectType; 5 bslbf
    byte ch = (byte) (bb.get(0) & 0xf8);
    // 3bits left.

    // samplingFrequencyIndex; 4 bslbf
    byte samplingFrequencyIndex = 0x04; //44100
    if (sampleRate == SrsCodecAudioSampleRate.R22050) {
      samplingFrequencyIndex = 0x07;
    } else if (sampleRate == SrsCodecAudioSampleRate.R11025) {
      samplingFrequencyIndex = 0x0a;
    } else if (sampleRate == SrsCodecAudioSampleRate.R32000) {
      samplingFrequencyIndex = 0x05;
    } else if (sampleRate == SrsCodecAudioSampleRate.R16000) {
      samplingFrequencyIndex = 0x08;
    }
    ch |= (samplingFrequencyIndex >> 1) & 0x07;
    audio_tag.put(ch, 2);

    ch = (byte) ((samplingFrequencyIndex << 7) & 0x80);
    // 7bits left.

    // channelConfiguration; 4 bslbf
    byte channelConfiguration = 1;
    if (achannel == 2) {
      channelConfiguration = 2;
    }
    ch |= (channelConfiguration << 3) & 0x78;
    // 3bits left.

    // GASpecificConfig(), page 451
    // 4.4.1 Decoder configuration (GASpecificConfig)
    // frameLengthFlag; 1 bslbf
    // dependsOnCoreCoder; 1 bslbf
    // extensionFlag; 1 bslbf
    audio_tag.put(ch, 3);

    aac_specific_config_got = true;
    aac_packet_type = 0; // 0 = AAC sequence header

    writeAdtsHeader(audio_tag.array(), 4);
    audio_tag.appendOffset(7);
  } else {
    bb.get(audio_tag.array(), 2, bi.size);
    audio_tag.appendOffset(bi.size + 2);
  }

  byte sound_format = 10; // AAC
  byte sound_type = 0; // 0 = Mono sound
  if (achannel == 2) {
    sound_type = 1; // 1 = Stereo sound
  }
  byte sound_size = 1; // 1 = 16-bit samples
  byte sound_rate = 3; // 44100, 22050, 11025
  if (sampleRate == 22050) {
    sound_rate = 2;
  } else if (sampleRate == 11025) {
    sound_rate = 1;
  }

  // for audio frame, there is 1 or 2 bytes header:
  //      1bytes, SoundFormat|SoundRate|SoundSize|SoundType
  //      1bytes, AACPacketType for SoundFormat == 10, 0 is sequence header.
  byte audio_header = (byte) (sound_type & 0x01);
  audio_header |= (sound_size << 1) & 0x02;
  audio_header |= (sound_rate << 2) & 0x0c;
  audio_header |= (sound_format << 4) & 0xf0;

  audio_tag.put(audio_header, 0);
  audio_tag.put(aac_packet_type, 1);

  writeRtmpPacket(SrsCodecFlvTag.Audio, dts, 0, aac_packet_type, audio_tag);
}
 
开发者ID:pedroSG94,项目名称:rtmp-rtsp-stream-client-java,代码行数:82,代码来源:SrsFlvMuxer.java


注:本文中的android.media.MediaCodec.BufferInfo方法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。