Android PCM->；AAC（编码器）——>；PCM（译码器）实时优化_Android_Pcm_Aac_Android Mediacodec_Audiorecord

Android PCM->；AAC（编码器）——>；PCM（译码器）实时优化

android

Android PCM->；AAC（编码器）——>；PCM（译码器）实时优化,android,pcm,aac,android-mediacodec,audiorecord,Android,Pcm,Aac,Android Mediacodec,Audiorecord,我正在努力实现 AudioRecord (MIC) -> PCM -> AAC Encoder AAC -> PCM Decode -> AudioTrack?? (SPEAKER) 使用Android 4.1+（API16）上的MediaCodec 首先，我通过MediaCodec成功地实现了PCM->AAC编码器（但不确定是否正确优化），如下所示 private boolean setEncoder(int rate) { encoder = Medi

我正在努力实现

AudioRecord (MIC) ->

PCM -> AAC Encoder
AAC -> PCM Decode

-> AudioTrack??  (SPEAKER)

使用Android 4.1+（API16）上的

MediaCodec

首先，我通过

MediaCodec

成功地实现了

PCM->AAC编码器

（但不确定是否正确优化），如下所示

private boolean setEncoder(int rate)
{
    encoder = MediaCodec.createEncoderByType("audio/mp4a-latm");
    MediaFormat format = new MediaFormat();
    format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm");
    format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1);
    format.setInteger(MediaFormat.KEY_SAMPLE_RATE, 44100);
    format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);//AAC-HE 64kbps
    format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE);
    encoder.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
    return true;
}

输入：PCM比特率=44100（Hz）x16（位）x1（单端）=705600位/秒
输出：AAC-HE比特率=64 x 1024（位）=65536位/s
因此，数据大小大约被压缩
x11
，我通过观察日志确认了这一点

录音记录器﹕ 读取4096字节

音频编码器﹕ 369字节编码

数据大小大约是压缩的
x11
，到目前为止还不错
现在，我有一个UDP服务器来接收编码数据，然后对其进行解码
解码器配置文件设置如下：

private boolean setDecoder(int rate) { decoder = MediaCodec.createDecoderByType("audio/mp4a-latm"); MediaFormat format = new MediaFormat(); format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm"); format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1); format.setInteger(MediaFormat.KEY_SAMPLE_RATE, 44100); format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);//AAC-HE 64kbps format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE); decoder.configure(format, null, null, 0); return true; }

IOudpPlayer = new Thread(new Runnable() { public void run() { SocketAddress sockAddress; String address; int len = 1024; byte[] buffer2 = new byte[len]; DatagramPacket packet; byte[] data; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; try { decoder.start(); isPlaying = true; while (isPlaying) { try { packet = new DatagramPacket(buffer2, len); ds.receive(packet); sockAddress = packet.getSocketAddress(); address = sockAddress.toString(); Log.d("UDP Receiver"," received !!! from " + address); data = new byte[packet.getLength()]; System.arraycopy(packet.getData(), packet.getOffset(), data, 0, packet.getLength()); Log.d("UDP Receiver", data.length + " bytes received"); //=========== inputBuffers = decoder.getInputBuffers(); outputBuffers = decoder.getOutputBuffers(); inputBufferIndex = decoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(data); decoder.queueInputBuffer(inputBufferIndex, 0, data.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); Log.d("AudioDecoder", outData.length + " bytes decoded"); decoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); } //=========== } catch (IOException e) { } } decoder.stop(); } catch (Exception e) { } } });
由于UDPserver数据包缓冲区大小为
1024

UDPserver﹕ 接收到1024字节

由于这是压缩的AAC数据，我希望解码的大小是
大约1024个
x11
，但实际结果是

音频解码器﹕ 已解码8192字节

大约是
x8
，我感觉有点不对劲
解码器代码如下所示：

private boolean setDecoder(int rate) { decoder = MediaCodec.createDecoderByType("audio/mp4a-latm"); MediaFormat format = new MediaFormat(); format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm"); format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1); format.setInteger(MediaFormat.KEY_SAMPLE_RATE, 44100); format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);//AAC-HE 64kbps format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE); decoder.configure(format, null, null, 0); return true; }

IOudpPlayer = new Thread(new Runnable() { public void run() { SocketAddress sockAddress; String address; int len = 1024; byte[] buffer2 = new byte[len]; DatagramPacket packet; byte[] data; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; try { decoder.start(); isPlaying = true; while (isPlaying) { try { packet = new DatagramPacket(buffer2, len); ds.receive(packet); sockAddress = packet.getSocketAddress(); address = sockAddress.toString(); Log.d("UDP Receiver"," received !!! from " + address); data = new byte[packet.getLength()]; System.arraycopy(packet.getData(), packet.getOffset(), data, 0, packet.getLength()); Log.d("UDP Receiver", data.length + " bytes received"); //=========== inputBuffers = decoder.getInputBuffers(); outputBuffers = decoder.getOutputBuffers(); inputBufferIndex = decoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(data); decoder.queueInputBuffer(inputBufferIndex, 0, data.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); Log.d("AudioDecoder", outData.length + " bytes decoded"); decoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); } //=========== } catch (IOException e) { } } decoder.stop(); } catch (Exception e) { } } });
完整代码：

还需要许可：

<uses-permission android:name="android.permission.INTERNET"></uses-permission> <uses-permission android:name="android.permission.RECORD_AUDIO"></uses-permission>

然而，结果是一样的
现在，我想，错误是：
W/SoftAAC2﹕ AAC解码器返回错误16388，替换为静默。
表示此解码器从第一个开始就完全失败。又是数据不可查找的问题。如果AAC解码器不能以这种方式处理流式数据，而只是添加一些头，那将非常令人失望。

UPDATE2:UDP接收器出错，因此进行了修改

现在，错误来了
W/SoftAAC2﹕ AAC解码器返回错误16388，替换为静默。
消失了
因此，它表明解码器至少可以正常工作
但是，这是1个周期的日志：

D/AudioRecoder﹕ 4096 bytes read D/AudioEncoder﹕ 360 bytes encoded D/UDP Receiver﹕ received !!! from /127.0.0.1:39000 D/UDP Receiver﹕ 360 bytes received D/AudioDecoder﹕ 8192 bytes decoded
PCM（4096）->AACencoded（360）->UDP-AAC（360）->（应该是）PCM（8192）
最终的结果是原PCM的2倍大小，但仍有问题

所以我的问题是

你能适当地优化我的示例代码以使其正常工作吗

使用
AudioTrack
API动态播放解码后的PCM原始数据是否正确，您能告诉我正确的方法吗？欢迎使用示例代码
多谢各位

另外，我的项目目标是Android4.1+（API16），我读过API18（Android 4.3+）上的东西更简单，但出于明显的兼容性原因，不幸的是，我不得不跳过MediaMuxer等。这里…
您的网络代码正在组合数据。你得到了369字节的压缩数据，但在接收端你得到了1024字节。这1024字节由两个完整帧和一个部分帧组成。这两个完整的帧再次解码为4096字节，总共8192字节。一旦您向解码器发送足够多的数据，剩余的部分帧可能会被解码，但您通常应该只向解码器发送整个帧

此外，
MediaCodec.dequeueOutputBuffer（）
不仅返回（正）缓冲区索引，还返回（负）状态代码。可能的代码之一是
MediaCodec.INFO\u OUTPUT\u FORMAT\u CHANGED
，这表示您需要调用
MediaCodec.getOutputFormat（）
，以获取音频数据的格式。即使输入是单声道，也可能会看到编解码器输出立体声。当您发布的代码收到这些状态代码之一时，它就会跳出循环。
自我回答，以下是我迄今为止所做的最大努力

package com.example.app; import android.app.Activity; import android.media.AudioManager; import android.media.MediaCodecInfo; import android.media.MediaFormat; import android.os.Bundle; import android.media.AudioFormat; import android.media.AudioRecord; import android.media.AudioTrack; import android.media.MediaCodec; import android.media.MediaRecorder.AudioSource; import android.util.Log; import java.io.IOException; import java.net.DatagramPacket; import java.net.DatagramSocket; import java.net.InetAddress; import java.net.SocketAddress; import java.net.SocketException; import java.nio.ByteBuffer; public class MainActivity extends Activity { private AudioRecord recorder; private AudioTrack player; private MediaCodec encoder; private MediaCodec decoder; private short audioFormat = AudioFormat.ENCODING_PCM_16BIT; private short channelConfig = AudioFormat.CHANNEL_IN_MONO; private int bufferSize; private boolean isRecording; private boolean isPlaying; private Thread IOrecorder; private Thread IOudpPlayer; private DatagramSocket ds; private final int localPort = 39000; @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.activity_main); IOrecorder = new Thread(new Runnable() { public void run() { int read; byte[] buffer1 = new byte[bufferSize]; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; DatagramPacket packet; try { encoder.start(); recorder.startRecording(); isRecording = true; while (isRecording) { read = recorder.read(buffer1, 0, bufferSize); // Log.d("AudioRecoder", read + " bytes read"); //------------------------ inputBuffers = encoder.getInputBuffers(); outputBuffers = encoder.getOutputBuffers(); inputBufferIndex = encoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(buffer1); encoder.queueInputBuffer(inputBufferIndex, 0, buffer1.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = encoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); // Log.d("AudioEncoder", outData.length + " bytes encoded"); //------------- packet = new DatagramPacket(outData, outData.length, InetAddress.getByName("127.0.0.1"), localPort); ds.send(packet); //------------ encoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = encoder.dequeueOutputBuffer(bufferInfo, 0); } // ----------------------; } encoder.stop(); recorder.stop(); } catch (Exception e) { e.printStackTrace(); } } }); IOudpPlayer = new Thread(new Runnable() { public void run() { SocketAddress sockAddress; String address; int len = 1024; byte[] buffer2 = new byte[len]; DatagramPacket packet; byte[] data; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; try { player.play(); decoder.start(); isPlaying = true; while (isPlaying) { try { packet = new DatagramPacket(buffer2, len); ds.receive(packet); sockAddress = packet.getSocketAddress(); address = sockAddress.toString(); // Log.d("UDP Receiver"," received !!! from " + address); data = new byte[packet.getLength()]; System.arraycopy(packet.getData(), packet.getOffset(), data, 0, packet.getLength()); // Log.d("UDP Receiver", data.length + " bytes received"); //=========== inputBuffers = decoder.getInputBuffers(); outputBuffers = decoder.getOutputBuffers(); inputBufferIndex = decoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(data); decoder.queueInputBuffer(inputBufferIndex, 0, data.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); // Log.d("AudioDecoder", outData.length + " bytes decoded"); player.write(outData, 0, outData.length); decoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); } //=========== } catch (IOException e) { } } decoder.stop(); player.stop(); } catch (Exception e) { } } }); //=========================================================== int rate = findAudioRecord(); if (rate != -1) { Log.v("=========media ", "ready: " + rate); Log.v("=========media channel ", "ready: " + channelConfig); boolean encoderReady = setEncoder(rate); Log.v("=========encoder ", "ready: " + encoderReady); if (encoderReady) { boolean decoderReady = setDecoder(rate); Log.v("=========decoder ", "ready: " + decoderReady); if (decoderReady) { Log.d("=======bufferSize========", "" + bufferSize); try { setPlayer(rate); ds = new DatagramSocket(localPort); IOudpPlayer.start(); IOrecorder.start(); } catch (SocketException e) { e.printStackTrace(); } } } } } protected void onDestroy() { recorder.release(); player.release(); encoder.release(); decoder.release(); } /* protected void onResume() { // isRecording = true; } protected void onPause() { isRecording = false; } */ private int findAudioRecord() { for (int rate : new int[]{44100}) { 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; } private boolean setEncoder(int rate) { encoder = MediaCodec.createEncoderByType("audio/mp4a-latm"); MediaFormat format = new MediaFormat(); format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm"); format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1); format.setInteger(MediaFormat.KEY_SAMPLE_RATE, rate); format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);//AAC-HE 64kbps format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE); encoder.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE); return true; } private boolean setDecoder(int rate) { decoder = MediaCodec.createDecoderByType("audio/mp4a-latm"); MediaFormat format = new MediaFormat(); format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm"); format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, 1); format.setInteger(MediaFormat.KEY_SAMPLE_RATE, rate); format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);//AAC-HE 64kbps format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectHE); decoder.configure(format, null, null, 0); return true; } private boolean setPlayer(int rate) { int bufferSizePlayer = AudioTrack.getMinBufferSize(rate, AudioFormat.CHANNEL_OUT_MONO, audioFormat); Log.d("====buffer Size player ", String.valueOf(bufferSizePlayer)); player= new AudioTrack(AudioManager.STREAM_MUSIC, rate, AudioFormat.CHANNEL_OUT_MONO, audioFormat, bufferSizePlayer, AudioTrack.MODE_STREAM); if (player.getState() == AudioTrack.STATE_INITIALIZED) { return true; } else { return false; } } }

在测试之后，我通过修改您的代码得出了以下结论：

package com.example.app; import android.app.Activity; import android.media.AudioManager; import android.media.MediaCodecInfo; import android.media.MediaFormat; import android.os.Bundle; import android.media.AudioFormat; import android.media.AudioRecord; import android.media.AudioTrack; import android.media.MediaCodec; import android.media.MediaRecorder.AudioSource; import android.util.Log; import java.io.IOException; import java.net.DatagramPacket; import java.net.DatagramSocket; import java.net.InetAddress; import java.net.SocketAddress; import java.net.SocketException; import java.nio.ByteBuffer; public class MainActivity extends Activity { private AudioRecord recorder; private AudioTrack player; private MediaCodec encoder; private MediaCodec decoder; private short audioFormat = AudioFormat.ENCODING_PCM_16BIT; private short channelConfig = AudioFormat.CHANNEL_IN_MONO; private int bufferSize; private boolean isRecording; private boolean isPlaying; private Thread IOrecorder; private Thread IOudpPlayer; private DatagramSocket ds; private final int localPort = 39000; @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.activity_main); IOrecorder = new Thread(new Runnable() { public void run() { int read; byte[] buffer1 = new byte[bufferSize]; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; DatagramPacket packet; try { encoder.start(); recorder.startRecording(); isRecording = true; while (isRecording) { read = recorder.read(buffer1, 0, bufferSize); // Log.d("AudioRecoder", read + " bytes read"); //------------------------ inputBuffers = encoder.getInputBuffers(); outputBuffers = encoder.getOutputBuffers(); inputBufferIndex = encoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(buffer1); encoder.queueInputBuffer(inputBufferIndex, 0, buffer1.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = encoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); // Log.d("AudioEncoder ", outData.length + " bytes encoded"); //------------- packet = new DatagramPacket(outData, outData.length, InetAddress.getByName("127.0.0.1"), localPort); ds.send(packet); //------------ encoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = encoder.dequeueOutputBuffer(bufferInfo, 0); } // ----------------------; } encoder.stop(); recorder.stop(); } catch (Exception e) { e.printStackTrace(); } } }); IOudpPlayer = new Thread(new Runnable() { public void run() { SocketAddress sockAddress; String address; int len = 2048 byte[] buffer2 = new byte[len]; DatagramPacket packet; byte[] data; ByteBuffer[] inputBuffers; ByteBuffer[] outputBuffers; ByteBuffer inputBuffer; ByteBuffer outputBuffer; MediaCodec.BufferInfo bufferInfo; int inputBufferIndex; int outputBufferIndex; byte[] outData; try { player.play(); decoder.start(); isPlaying = true; while (isPlaying) { try { packet = new DatagramPacket(buffer2, len); ds.receive(packet); sockAddress = packet.getSocketAddress(); address = sockAddress.toString(); // Log.d("UDP Receiver"," received !!! from " + address); data = new byte[packet.getLength()]; System.arraycopy(packet.getData(), packet.getOffset(), data, 0, packet.getLength()); // Log.d("UDP Receiver", data.length + " bytes received"); //=========== inputBuffers = decoder.getInputBuffers(); outputBuffers = decoder.getOutputBuffers(); inputBufferIndex = decoder.dequeueInputBuffer(-1); if (inputBufferIndex >= 0) { inputBuffer = inputBuffers[inputBufferIndex]; inputBuffer.clear(); inputBuffer.put(data); decoder.queueInputBuffer(inputBufferIndex, 0, data.length, 0, 0); } bufferInfo = new MediaCodec.BufferInfo(); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0); while (outputBufferIndex >= 0) { outputBuffer = outputBuffers[outputBufferIndex]; outputBuffer.position(bufferInfo.offset); outputBuffer.limit(bufferInfo.offset + bufferInfo.size); outData = new byte[bufferInfo.size]; outputBuffer.get(outData); // Log.d("AudioDecoder", outData.length + " bytes decoded"); player.write(outData, 0, outData.length); decoder.releaseOutputBuffer(outputBufferIndex, false); outputBufferIndex = decoder.dequeueOutputBuffer(bufferInfo, 0..

我已经用你的资源测试过了。有几点

比特率是K的自然数，但不是计算机K。 64k=64000，但不是64*1024

不建议编写共享某些变量的长代码。 A.将编码器线程和解码器线程分为2个独立类。 B.数据集邮是发送方和接收方共享的，这不好

枚举音频格式需要更多值。 i、 e.抽样率应从8000、11025、22050、44100中选取

我尝试了上面的代码，但它没有正常工作。我在解码输出中注入了大量的静默。问题是没有为解码器设置正确的“csd”值
因此，如果您在日志中看到“静默”或解码器抛出错误，请确保您已将以下内容添加到媒体解码器格式中

int profile = 2; //AAC LC int freqIdx = 11; //8KHz int chanCfg = 1; //Mono ByteBuffer csd = ByteBuffer.allocate(2); csd.put(0, (byte) (profile << 3 | freqIdx >> 1)); csd.put(1, (byte)((freqIdx & 0x01) << 7 | chanCfg << 3)); mediaFormat.setByteBuffer("csd-0", csd);

int profile=2//AAC信用证 int-freqIdx=11//8KHz int-chanCfg=1//单声道 ByteBuffer csd=ByteBuffer.allocate（2）； csd.put（0，（字节）（profile>1））； csd.put（1，（字节）（（freqIdx&0x01）D/audiorecord﹕ 读取4096字节音频编码器﹕ 360字节编码 D/UDP接收机﹕ 收到！！！从/127.0.0.1:39000 D/UDP接收机﹕ 接收到360字节音频解码器﹕ 已解码8192字节这是因为acc解码器始终解码到立体声通道，即使编码数据是单声道。因此，如果您的编码端设置为立体声通道，则如下所示：音频记录器﹕ 8192字节读取音频编码器﹕ 360字节编码 D/UDP接收机﹕ 收到！！！从/127.0.0.1:39000 D/UDP接收机﹕ 接收到360字节音频解码器﹕ 8192字节解码我不是这方面的专家，但我会假设每个数据包中都有额外的报头信息控制数据，使得原始有效负载小于1024，因此8192可能是合理的。更有趣的是音频听起来像什么？压缩后是否可以播放？是否可以播放