当前位置: 首页>>代码示例>>Java>>正文

Java AACLATMPacketizer类代码示例

本文整理汇总了Java中net.majorkernelpanic.streaming.rtp.AACLATMPacketizer类的典型用法代码示例。如果您正苦于以下问题：Java AACLATMPacketizer类的具体用法？Java AACLATMPacketizer怎么用？Java AACLATMPacketizer使用的例子？那么恭喜您, 这里精选的类代码示例或许可以为您提供帮助。


AACLATMPacketizer类属于net.majorkernelpanic.streaming.rtp包，在下文中一共展示了AACLATMPacketizer类的3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: encodeWithMediaCodec

import net.majorkernelpanic.streaming.rtp.AACLATMPacketizer; //导入依赖的package包/类
@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.setDestination(mDestination, mRtpPort, mRtcpPort);
	mPacketizer.setInputStream(inputStream);
	mPacketizer.start();

	mStreaming = true;

}
开发者ID:ghazi94，项目名称:Android_CCTV，代码行数:59，代码来源:AACStream.java

示例2: encodeWithMediaCodec

import net.majorkernelpanic.streaming.rtp.AACLATMPacketizer; //导入依赖的package包/类
@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);
	try {
		mMediaCodec = MediaCodec.createEncoderByType("audio/mp4a-latm");
	} catch (IOException e){
		Log.d(TAG, "Unable to instantiate a decoder, trying Google one...");
		mMediaCodec = MediaCodec.createByCodecName("OMX.google.aac.encoder");
	}
	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;

}
开发者ID:hypeapps，项目名称:Endoscope，代码行数:63，代码来源:AACStream.java

示例3: encodeWithMediaCodec

import net.majorkernelpanic.streaming.rtp.AACLATMPacketizer; //导入依赖的package包/类
@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;

}
开发者ID:quanhua92，项目名称:libstreaming_android_studio，代码行数:58，代码来源:AACStream.java


注：本文中的net.majorkernelpanic.streaming.rtp.AACLATMPacketizer类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。