There's a funky thing going on with Xuggler where it doesn't always allow you to set the sample rate of IAudioSamples
. You'll need to use an IAudioResampler
.
Took me a while to figure this out. This post by Marty helped a lot, though his code is outdated now.
Here's how you fix it.
.
Before encoding
I'm assuming here that audio input has been properly set up, resulting in an IStreamCoder
called audioCoder
.
After that's done, you are probably initiating an IMediaWriter
and adding an audio stream like so:
final IMediaWriter oggWriter = ToolFactory.makeWriter(oggOutputFile);
// Using stream 1 'cause there is also a video stream.
// For an audio only file you should use stream 0.
oggWriter.addAudioStream(1, 1, ICodec.ID.CODEC_ID_VORBIS,
audioCoder.getChannels(), audioCoder.getSampleRate());
Now create an IAudioResampler
:
IAudioResampler oggResampler = IAudioResampler.make(audioCoder.getChannels(),
audioCoder.getChannels(),
audioCoder.getSampleRate(),
audioCoder.getSampleRate(),
IAudioSamples.Format.FMT_FLT,
audioCoder.getSampleFormat());
And tell your IMediaWriter
to update to its sample format:
// The stream 1 here is consistent with the stream we added earlier.
oggWriter.getContainer().getStream(1).getStreamCoder().
setSampleFormat(IAudioSamples.Format.FMT_FLT);
.
During encoding
You are currently probably initiating an IAudioSamples
and filling it with audio data, like so:
IAudioSamples audioSample = IAudioSamples.make(512, audioCoder.getChannels(),
audioCoder.getSampleFormat());
int bytesDecoded = audioCoder.decodeAudio(audioSample, packet, offset);
Now initiate an IAudioSamples for our resampled data:
IAudioSamples vorbisSample = IAudioSamples.make(512, audioCoder.getChannels(),
IAudioSamples.Format.FMT_FLT);
Finally, resample the audio data and write the result:
oggResampler.resample(vorbisSample, audioSample, 0);
oggWriter.encodeAudio(1, vorbisSample);
.
Final thought
Just a hint to get your output files to play well:
- If you use audio and video within the same container, then audio and video data packets should be written in such an order that the timestamp of each data packet is higher than that of the previous data packet. So you are almost certainly going to need some kind of buffering mechanism that alternates writing audio and video.