JavaのXugglerでaac/wav/wmaオーディオファイルをmp3に変換しようとしています。
残念ながら、品質が大幅に低下しています。入力ファイルのサイズは約7MBで、出力ファイルのサイズはわずか1.5MBです。
サンプルレートは44100Hzに設定されていますが、他に設定するパラメータはありますか？
ご回答ありがとうございます。
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);
}
}