AAC streamed byte packets and recordSamples

I have a use case to get packets from a streaming server, from an RTMP encoder. And transcode to Ogg/Opus using recordSamples. It's similar to the microphone examples but different. Giving the recorder the raw AAC bytes produces a noisy output. Using the frame grabber to be the rtmp client works.

Does the AAC packets need to be decoded to PCM first to be usable ? Is there a way to do this without passing them to the framegrabber or can they be sent to a frame grabber using PipedInputStream ?

It's not a common use case it seems I only see microphone inputs that is PCM already. The server might have a utility already to decode the AAC packets to PCM to be consumable by the recorder.

Example of what I am trying to do right now but the output has to go to an outputstream for sending bytes to a websocket service.

FFmpegFrameRecorder recorder = new FFmpegFrameRecorder("D:\\test.ogg", 1);
	    	recorder.setFormat("ogg");
	    	recorder.setAudioCodec(AV_CODEC_ID_OPUS);
	    	recorder.setSampleFormat(AV_SAMPLE_FMT_FLT);
	    	recorder.setAudioOption("vbr", "constrained");
	        recorder.setAudioBitrate(24000);
	        recorder.setSampleRate(16000);
	        recorder.setAudioChannels(1);	
 recorder.start();

Then in the packet capture method. I dont know the method yet to get the samplerate from the input packets. packet.getData is an AAC packet.

byte[] data = packet.getData();
ByteBuffer buffer = ByteBuffer.wrap(data);
recorder.recordSamples(48000, 2, buffer);