Not able to receive streamed audio using FFmpegFrameGrabber

Hi, I am using FFMpegRecorer to stream audio with "wav" format from one java application and using FFMpegGrabber in another java application to receive the stream but I am getting the following error while starting the FFMpegGrabber,

Error: [wav @ 00000166fe631c00] invalid start code [33][25]EB in RIFF header Error: [wav @ 00000166fe631c00] invalid start code [33][25]Eq in RIFF header avformat_open_input() error -1094995529: Could not open input "udp://127.0.0.1:1234". (Has setFormat() been called?) (For more details, make sure FFmpegLogCallback.set() has been called.)

What did I do wrong in the following code?

Code I am using on the streaming side:

public void stream() {
        AudioFormat      audioFormat = new AudioFormat(44100.0F, 16, 2, true, false);
        final int sampleRate = (int) audioFormat.getSampleRate();
        final int numChannels = audioFormat.getChannels();
        final int audioBufferSize = sampleRate * numChannels;
        final byte[] audioBytes = new byte[audioBufferSize];
        dataLineInfo = new DataLine.Info(TargetDataLine.class, audioFormat);
        micDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
        micDataLine.open(audioFormat);
        micDataLine.start();
        FFmpegFrameRecorder  frameRecorderForStreamAudio = new FFmpegFrameRecorder("udp://127.0.01:1234", 2);
        frameRecorderForStreamAudio.setInterleaved(true);
        frameRecorderForStreamAudio.setFormat("wav");
        frameRecorderForStreamAudio.setAudioCodec(avcodec.AV_CODEC_ID_AAC);
        frameRecorderForStreamAudio.setAudioOption("preset", "veryfast");
        frameRecorderForStreamAudio.setAudioOption("tune", "zerolatency");
        frameRecorderForStreamAudio.setOption("force_key_frames", "0:00:01");
        frameRecorderForStreamAudio.setGopSize(1);
        frameRecorderForStreamAudio.start();
             timer = new Timer(true);
             timerTask = new TimerTask() {
             @Override
             public void run() {
                int nBytesRead = micDataLine.read(audioBytes, 0, micDataLine.available());
                int nSamplesRead = nBytesRead / 2;
                short[] samples = new short[nSamplesRead];
                ByteBuffer.wrap(audioBytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(samples);
                ShortBuffer sBuff = ShortBuffer.wrap(samples, 0, nSamplesRead);
                if (!isStreamingPaused) {
                    try {
           
                        frameRecorderForStreamAudio.recordSamples(sampleRate, numChannels, sBuff);

                    } catch (FFmpegFrameRecorder.Exception e) {
                        e.printStackTrace();
                    }
                }
            }
        };
        timer.schedule(timerTask, 1000, (long) 1000 / Constants.VIDEO_FRAME_RATE);
    }

Code I am using on the receiver end,

    class PlayAudio implements Runnable {
        protected volatile boolean runnable = false;

        @Override
        public void run() {
            synchronized (this) {
                try {
                    FFmpegFrameGrabberframeGrabberForReceiveAudio = new FFmpegFrameGrabber("udp://127.0.0.1:1234");
                    frameGrabberForReceiveAudio.setFormat("wav");
                    frameGrabberForReceiveAudio.setAudioCodec(avcodec.AV_CODEC_ID_AAC);
                    System.setProperty("org.bytedeco.javacpp.logger.debug", "true");
                    FFmpegLogCallback.set();
                    **frameGrabberForReceiveAudio.start();**
              

              } catch (FFmpegFrameGrabber.Exception e) {
                    logger.log(Level.SEVERE, e.getMessage());
                }
                if (frameGrabberForReceiveAudio.getAudioChannels() > 0) {
                    final AudioFormat audioFormat = new AudioFormat(frameGrabberForReceiveAudio.getSampleRate(), 16, 
                    frameGrabberForReceiveAudio.getAudioChannels(), true, true);
                    final DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
                    try {
                        speakerDataLine = (SourceDataLine) AudioSystem.getLine(info);
                        speakerDataLine.open(audioFormat);
                        speakerDataLine.start();
                    } catch (LineUnavailableException e) {
                        throw new RuntimeException(e);
                    }
                    try {
                        Frame f = frameGrabberForReceiveAudio.grab();
                        logger.info("Started receiving audio stream");
                        hideBtnVideoLoadingIndicator();
                        indicateMediaLive();
                        showBtnPauseStreamedMedia();
                        while (frameGrabberForReceiveAudio.grab() != null) {
                            ShortBuffer channelSamplesShortBuffer = (ShortBuffer) f.samples[0];
                            channelSamplesShortBuffer.rewind();
                            final ByteBuffer outBuffer = ByteBuffer.allocate(channelSamplesShortBuffer.capacity() * 2);
                            for (int i = 0; i < channelSamplesShortBuffer.capacity(); i++) {
                                short val = channelSamplesShortBuffer.get(i);
                                outBuffer.putShort(val);
                            }
                            if (!isStreamedAudioPaused) {
                                speakerDataLine.write(outBuffer.array(), 0, outBuffer.capacity());
                            }
//                        logger.info("Playing");
                        }
                    } catch (FFmpegFrameGrabber.Exception e) {
                        logger.info(e.getMessage());
                    }
                }
            }
        }
    }

I also tried to play the audio with ffplay udp://127.0.0.1:1234 from command line there I got the following errors,

[aac @ 0000024d68f7e5c0] Reserved bit set. [aac @ 0000024d68f7e5c0] Prediction is not allowed in AAC-LC. [aac @ 0000024d68f7e5c0] Number of scalefactor bands in group (43) exceeds limit (40). [aac @ 0000024d68f7e5c0] Reserved bit set. 0KB sq= 0B f=0/0 [aac @ 0000024d68f7e5c0] channel element 3.12 is not allocated [aac @ 0000024d68f7e5c0] channel element 2.9 is not allocated=0/0 [aac @ 0000024d68f7e5c0] Prediction is not allowed in AAC-LC.=0/0 [aac @ 0000024d68f7e5c0] channel element 3.1 is not allocated [aac @ 0000024d68f7e5c0] channel element 3.2 is not allocated=0/0 [aac @ 0000024d68f7e5c0] channel element 3.4 is not allocated=0/0 [aac @ 0000024d68f7e5c0] Number of bands (17) exceeds limit (12). [aac @ 0000024d68f7e5c0] channel element 2.2 is not allocated=0/0 [aac @ 0000024d68f7e5c0] Number of bands (25) exceeds limit (23). [aac @ 0000024d68f7e5c0] Sample rate index in program config element does not match the sample rate index configured by the container. [aac @ 0000024d68f7e5c0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0000024d68f7e5c0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org) [aac @ 0000024d68f7e5c0] Sample rate index in program config element does not match the sample rate index configured by the container.