我正在编写一个核心Java应用程序(JDK 11)，该应用程序应该记录音频和视频。
经过对各种库的广泛尝试和错误，我设法使用不推荐使用的Xuggler库来同时运行这两个库。
但是，以不错的质量录制音频仍然是一个问题。
我设法将录音作为short []样本进行编码，但是由于某些原因，它们被TargetDataLine在振幅127处截断了。我可以通过将它们乘以一个系数来增加编码量，但是高于127的任何录音细节都是迷失了声音。
就是事实发生之后(我失去了声音或正常语音)，我可以将麦克风扩音并放大。
不幸的是，我无法在Java中控制FloatControl.Type.MASTER_GAIN，因为AudioSystem似乎不支持任何控件类型(如果这样做可能会解决问题)；
问题:
如何从TargetDataLine捕获完整的声音/采样幅度，而在127处不被截断？
研究指出了以下有用的线索:
How to get Audio for encoding using Xuggler
How to set volume of a SourceDataLine in Java
Java algorithm for normalizing audio
Xuggler encoding and muxing
这是我的代码:
private static void startRecordingVideo() {
// total duration of the media
long duration = DEFAULT_TIME_UNIT.convert(1, SECONDS);
// video parameters
//Dimension size = WebcamResolution.QVGA.getSize();
//webcam.setViewSize(size);
BufferedImage img = webCamImageStream.get();
final int videoStreamIndex = 0;
final int videoStreamId = 0;
final long frameRate = DEFAULT_TIME_UNIT.convert(2, MILLISECONDS);
// audio parameters
TargetDataLine mic = null;
final int audioStreamIndex = 1;
final int audioStreamId = 0;
final int channelCount = 2; //1 mono 2Stereo
final int sampleRate = 44100; // Hz
final int sampleSizeInBits = 16; // bit in sample
final int frameSizeInByte = 4;
final int sampleCount = 588; //CD standard (588 lines per frame)
// the clock time of the next frame
long nextFrameTime = 0;
// the total number of audio samples
long totalSampleCount = 0;
// create a media writer and specify the output file
final IMediaWriter writer = ToolFactory.makeWriter("capture.mp4");
// add the video stream
writer.addVideoStream(videoStreamIndex, videoStreamId,
img.getWidth(), img.getHeight());
// add the audio stream
writer.addAudioStream(audioStreamIndex, audioStreamId,
channelCount, sampleRate);
//define audio format
AudioFormat audioFormat = new AudioFormat(
AudioFormat.Encoding.PCM_SIGNED,
sampleRate,
sampleSizeInBits,
channelCount,
frameSizeInByte,
sampleRate,
true);
DataLine.Info info = new DataLine.Info(TargetDataLine.class, audioFormat);
AudioInputStream audioInputStream = null;
try {
mic = (TargetDataLine) AudioSystem.getLine(info);
//mic.open();
mic.open(audioFormat, mic.getBufferSize());
// Adjust the volume on the output line.
if (mic.isControlSupported(FloatControl.Type.MASTER_GAIN)) {
FloatControl gain = (FloatControl) mic.getControl(FloatControl.Type.MASTER_GAIN);
gain.setValue(-10.0f); // attempt to Reduce volume by 10 dB.
}else {
System.out.println("Not supported in my case :'( ");
}
mic.start();
audioInputStream = new AudioInputStream(mic);
} catch (Exception e) {
e.printStackTrace();
}
// loop through clock time, which starts at zero and increases based
// on the total number of samples created thus far
long start = System.currentTimeMillis();
//duration = frameRate;
recordingVideo = true;
updateUI("Recording");
System.out.println("Audio Buffer size : " + mic.getBufferSize());
coverImage = webCamImageStream.get();
int frameCount = 0;
//IGNOR Complexity of for Loop*******************************************************************
for (long clock = 0; clock < duration; clock = IAudioSamples.samplesToDefaultPts(totalSampleCount, sampleRate)){
// while the clock time exceeds the time of the next video frame,
// get and encode the next video frame
while (frameCount * clock >= nextFrameTime) {
BufferedImage image = webCamImageStream.get();
IConverter converter = ConverterFactory.createConverter(image, IPixelFormat.Type.YUV420P);
IVideoPicture frame = converter.toPicture(image, (System.currentTimeMillis() - start) * 1000);
writer.encodeVideo(videoStreamIndex, frame);
nextFrameTime += frameRate;
}
//##################################### Audio Recording section #######################################
int factor = 2;
byte[] audioBytes = new byte[mic.getBufferSize() ]; // best size?
int numBytesRead = 0;
try {
numBytesRead = audioInputStream.read(audioBytes, 0, audioBytes.length);
//error is probably here as it is only reading up to 127
} catch (IOException e) {
numBytesRead = mic.read(audioBytes, 0, audioBytes.length);
e.printStackTrace();
}
mic.flush();
// max for normalizing
short rawMax = Short.MIN_VALUE;
for (int i = 0; i < numBytesRead; ++i) {
short value = audioBytes[i];
rawMax = (short) Math.max(rawMax, value);
}
//127 is max input amplitude (microphone could go higher but its cut off) ###############################
//values at and over 127 are static noises
System.out.println("MAX = " +rawMax );
// convert to signed shorts representing samples
int volumeGainfactor = 2;
int numSamplesRead = numBytesRead / factor;
short[] audioSamples = new short[ numSamplesRead ];
if (audioFormat.isBigEndian()) {
for (int i = 0; i < numSamplesRead; i++) {
audioSamples[i] = (short)((audioBytes[factor*i] << 8) | audioBytes[factor*i + 1]);
}
}
else {
for (int i = 0; i < numSamplesRead; i++) {
audioSamples[i] = (short)(((audioBytes[factor*i + 1] ) << 8) |(audioBytes[factor*i])) ;
//normalization -> does not help (issue lies in Max read value)
//short targetMax = 127; //maximum volume
//Normalization method
/*
double maxReduce = 1 - targetMax/(double)rawMax;
int abs = Math.abs(audioSamples[i]);
double factor1 = (maxReduce * abs/(double)rawMax);
audioSamples[i] = (short) Math.round((1 - factor1) * audioSamples[i]);
*/
//https://stackoverflow.com/questions/12469361/java-algorithm-for-normalizing-audio
}
}
//##################################### END Audio Recording Section #####################################
writer.encodeAudio(audioStreamIndex, audioSamples, clock,
DEFAULT_TIME_UNIT);
//extend duration if video is not terminated
if(!recordingVideo) {break;}
else {duration += 22675;} //should never catch up to duration
// 22675 = IAudioSamples.samplesToDefaultPts(588, sampleRate)
//totalSampleCount += sampleCount;
totalSampleCount = sampleCount;
frameCount++;
}
// manually close the writer
writer.close();
mic.close();
}
MAX = 48 (is recorded)
MAX = 127 (is static noise)
最佳答案
好的，看来我设法通过反复试验来解决此问题，并发布了这篇文章:
reading wav/wave file into short[] array
问题在于将byte [](起源)转换为short []。
AudioFormat audioFormat = new AudioFormat(
AudioFormat.Encoding.PCM_SIGNED,
sampleRate,
sampleSizeInBits,
channelCount,
frameSizeInByte,
sampleRate,
false);`
int factor = 2;
byte[] audioBytes = new byte[mic.getBufferSize() ];
int numBytesRead = 0;
numBytesRead = audioInputStream.read(audioBytes, 0, audioBytes.length);
mic.flush();
// convert to signed shorts representing samples
int volumeGainfactor = 2;
int numSamplesRead = numBytesRead / factor;
short[] audioSamples = new short[ numSamplesRead ];
if (audioFormat.isBigEndian()) {
for (int i = 0; i < numSamplesRead; i++) {
//BigEndian Conversion not working
audioSamples[i] = (short)((audioBytes[factor*i] << 8) | audioBytes[factor*i + 1]);
}
}
else {
for (int i = 0; i < numSamplesRead; i++) {
____________________________________________________ ISSUE WAS HERE __________________________________________
audioSamples[i] = ( (short)( ( audioBytes[i*2] & 0xff )|( audioBytes[i*2 + 1] << 8 )) );
____________________________________________________________________________________________________
}
}
关于java - Java麦克风TargetDataLine灵敏度/最大输入振幅，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/64062759/