Java에서 Xuggler로 aac/wav/wma 오디오 파일을 mp3로 변환하려고합니다.Xuggler로 오디오 변환
불행히도, 나는 품질이 크게 떨어졌습니다. 내 입력 파일 크기는 약 7MB이고 출력 파일 크기는 1,5MB입니다.
샘플 속도가 44100Hz로 설정되었으므로 설정할 다른 매개 변수가 있습니까?
답장을 보내 주셔서 감사합니다.
if (args.length <= 1)
throw new IllegalArgumentException("must pass an input filename and output filename as argument");
IMediaWriter writer = ToolFactory.makeWriter(args[1]);
String filename = args[0];
// Create a Xuggler container object
IContainer container = IContainer.make();
// Open up the container
if (container.open(filename, IContainer.Type.READ, null) < 0)
throw new IllegalArgumentException("could not open file: " + filename);
// query how many streams the call to open found
int numStreams = container.getNumStreams();
// and iterate through the streams to find the first audio stream
int audioStreamId = -1;
IStreamCoder audioCoder = null;
for(int i = 0; i < numStreams; i++)
{
// Find the stream object
IStream stream = container.getStream(i);
// Get the pre-configured decoder that can decode this stream;
IStreamCoder coder = stream.getStreamCoder();
if (coder.getCodecType() == ICodec.Type.CODEC_TYPE_AUDIO)
{
audioStreamId = i;
audioCoder = coder;
audioCoder.setBitRate(container.getBitRate());
break;
}
}
if (audioStreamId == -1)
throw new RuntimeException("could not find audio stream in container: "+filename);
/* We read only AAC file for the moment */
if(audioCoder.getCodecID() != ICodec.ID.CODEC_ID_AAC
&& audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WAVPACK
&& audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAV1
&& audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAV2
&& audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAPRO
&& audioCoder.getCodecID() != ICodec.ID.CODEC_ID_WMAVOICE)
{
System.out.println("Read only AAC, WAV or WMA files");
System.exit(1);
}
audioCoder.setSampleFormat(IAudioSamples.Format.FMT_S16);
/*
* Now we have found the audio stream in this file. Let's open up our decoder so it can
* do work.
*/
if (audioCoder.open() < 0)
throw new RuntimeException("could not open audio decoder for container: "+filename);
int streamIndex = writer.addAudioStream(0, 0, audioCoder.getChannels(), audioCoder.getSampleRate());
System.out.println("audio Frame size : "+audioCoder.getAudioFrameSize());
/*
* Now, we start walking through the container looking at each packet.
*/
IPacket packet = IPacket.make();
while(container.readNextPacket(packet) >= 0)
{
/*
* Now we have a packet, let's see if it belongs to our audio stream
*/
if (packet.getStreamIndex() == audioStreamId)
{
/*
* We allocate a set of samples with the same number of channels as the
* coder tells us is in this buffer.
*
* We also pass in a buffer size (1024 in our example), although Xuggler
* will probably allocate more space than just the 1024 (it's not important why).
*/
IAudioSamples samples = IAudioSamples.make(512, audioCoder.getChannels(),IAudioSamples.Format.FMT_S16);
/*
* A packet can actually contain multiple sets of samples (or frames of samples
* in audio-decoding speak). So, we may need to call decode audio multiple
* times at different offsets in the packet's data. We capture that here.
*/
int offset = 0;
/*
* Keep going until we've processed all data
*/
while(offset < packet.getSize())
{
int bytesDecoded = audioCoder.decodeAudio(samples, packet, offset);
if (bytesDecoded < 0)
throw new RuntimeException("got error decoding audio in: " + filename);
offset += bytesDecoded;
/*
* Some decoder will consume data in a packet, but will not be able to construct
* a full set of samples yet. Therefore you should always check if you
* got a complete set of samples from the decoder
*/
if (samples.isComplete())
{
writer.encodeAudio(streamIndex, samples);
}
}
}
else
{
/*
* This packet isn't part of our audio stream, so we just silently drop it.
*/
do {} while(false);
}
}
을 나는 비슷한 일을했고 나는, 1,024분의 512 샘플 크기를 가지고 몇 가지 테스트 후, 나는 4096로 전환 및 오디오 품질이 좋은, 그리고 문제를 많이했다 glitces가 없습니다 (512/1024와 같이) –