I am trying to convert an audio file with the header
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
I want to transcode this file to mp3 format. I have following code snippet but its not working well. I have written it using XUGGLER code snippet for transcoding audio and video.
Audio decoder is
audioDecoder = IStreamCoder.make(IStreamCoder.Direction.DECODING, ICodec.findDecodingCodec(ICodec.ID.CODEC_ID_PCM_S16LE));
audioDecoder.setSampleRate(44100);
audioDecoder.setBitRate(176400);
audioDecoder.setChannels(2);
audioDecoder.setTimeBase(IRational.make(1,1000));
if (audioDecoder.open(IMetaData.make(), IMetaData.make()) < 0)
return false;
return true;
Audio encoder is
outContainer = IContainer.make();
outContainerFormat = IContainerFormat.make();
outContainerFormat.setOutputFormat("mp3", urlOut, null);
int retVal = outContainer.open(urlOut, IContainer.Type.WRITE, outContainerFormat);
if (retVal < 0) {
System.out.println("Could not open output container");
return false;
}
outAudioCoder = IStreamCoder.make(IStreamCoder.Direction.ENCODING, ICodec.findEncodingCodec(ICodec.ID.CODEC_ID_MP3));
outAudioStream = outContainer.addNewStream(outAudioCoder);
outAudioCoder.setSampleRate(new Integer(44100));
outAudioCoder.setChannels(2);
retVal = outAudioCoder.open(IMetaData.make(), IMetaData.make());
if (retVal < 0) {
System.out.println("Could not open audio coder");
return false;
}
retVal = outContainer.writeHeader();
if (retVal < 0) {
System.out.println("Could not write output FLV header: ");
return false;
}
return true;
And here is encode method where i send packets of 32 byte to transcode
public void encode(byte[] audioFrame){
//duration of 1 video frame
long lastVideoPts = 0;
IPacket packet_out = IPacket.make();
int lastPos = 0;
int lastPos_out = 0;
IAudioSamples audioSamples = IAudioSamples.make(48000, audioDecoder.getChannels());
IAudioSamples audioSamples_resampled = IAudioSamples.make(48000, audioDecoder.getChannels());
//we always have 32 bytes/sample
int pos = 0;
int audioFrameLength = audioFrame.length;
int audioFrameCnt = 1;
iBuffer = IBuffer.make(null, audioFrame, 0, audioFrameLength);
IPacket packet = IPacket.make(iBuffer);
//packet.setKeyPacket(true);
packet.setTimeBase(IRational.make(1,1000));
packet.setDuration(20);
packet.setDts(audioFrameCnt*20);
packet.setPts(audioFrameCnt*20);
packet.setStreamIndex(1);
packet.setPosition(lastPos);
lastPos+=audioFrameLength;
int pksz = packet.getSize();
packet.setComplete(true, pksz);
/*
* A packet can actually contain multiple samples
*/
int offset = 0;
int retVal;
while(offset < packet.getSize())
{
int bytesDecoded = audioDecoder.decodeAudio(audioSamples, packet, offset);
if (bytesDecoded < 0)
throw new RuntimeException("got error decoding audio ");
offset += bytesDecoded;
if (audioSamples.isComplete())
{
int samplesConsumed = 0;
while (samplesConsumed < audioSamples.getNumSamples()) {
retVal = outAudioCoder.encodeAudio(packet_out, audioSamples, samplesConsumed);
if (retVal <= 0)
throw new RuntimeException("Could not encode audio");
samplesConsumed += retVal;
if (packet_out.isComplete()) {
packet_out.setPosition(lastPos_out);
packet_out.setStreamIndex(1);
lastPos_out+=packet_out.getSize();
retVal = outContainer.writePacket(packet_out);
if(retVal < 0){
throw new RuntimeException("Could not write data packet");
}
}
}
}
}
}
I get an output file but it doesnt get played. I have very little experience of audio encoding and sampling. Thanks in advance.
Finally I am able to live transcode a pcm stream to mp3 stream.
There were couple of issues in the code :
I was trying to transcode only one thing i.e audio and the code snippet was transcoding audio as well as video so there was an issue in setting stream index.
packet_out.setStreamIndex(1); packet_out.setStreamIndex(0)
Second thing was calculations ffmpeg guid
channel * bits * sampling rate = bit rate
This thing was miscalculated at my end.
Number of samples in audio samples depends upon your sampling rate. That was wrong in my code.
NOTE : this is a pretty old code
byte[] data = new byte[418];
public void encode(byte[] audioFrame) {
IPacket packet_out = IPacket.make();
int lastPos_out = 0;
IAudioSamples audioSamples = IAudioSamples.make(11025, audioDecoder.getChannels());
//IAudioSamples audioSamples_resampled = IAudioSamples.make(48000, audioDecoder.getChannels());
//we always have 32 bytes/sample
int pos = 0;
int audioFrameLength = audioFrame.length;
int audioFrameCnt = 1;
iBuffer = IBuffer.make(null, audioFrame, 0, audioFrameLength);
IPacket packet = IPacket.make(iBuffer);
//packet.setKeyPacket(true);
packet.setTimeBase(IRational.make(1, 1000));
packet.setStreamIndex(0);
int pksz = packet.getSize();
packet.setComplete(true, pksz);
/*
* A packet can actually contain multiple samples
*/
int offset = 0;
int retVal;
while (offset < packet.getSize()) {
int bytesDecoded = audioDecoder.decodeAudio(audioSamples, packet, offset);
if (bytesDecoded < 0)
throw new RuntimeException("got error decoding audio ");
offset += bytesDecoded;
if (audioSamples.isComplete()) {
/*audioResampler.resample(audioSamples_resampled, audioSamples, audioSamples.getNumSamples());
audioSamples_resampled.setPts(Global.NO_PTS);*/
int samplesConsumed = 0;
while (samplesConsumed < audioSamples.getNumSamples()) {
retVal = outAudioCoder.encodeAudio(packet_out, audioSamples, samplesConsumed);
if (retVal <= 0)
throw new RuntimeException("Could not encode audio");
samplesConsumed += retVal;
if (packet_out.isComplete()) {
packet_out.setPosition(lastPos_out);
packet_out.setStreamIndex(0);
lastPos_out += packet_out.getSize();
System.out.println("size" + packet_out.getSize());
packet_out.getByteBuffer().get(data,0,packet_out.getSize());
try {
fo.write(data);
} catch (IOException e) {
e.printStackTrace(); //To change body of catch statement use File | Settings | File Templates.
}
packet_out.reset();
//retVal = outContainer.writePacket(packet_out);
if (retVal < 0) {
throw new RuntimeException("Could not write data packet");
}
}
}
}
}
}