I am trying to play an mp3 from an internet source while streaming in the same time. I've writen the code below and it throws the exception WaveBadFormat calling WaveOut at the player.Init() line. How should I fix this?
string testLink = "http://users.skynet.be/fa046054/home/P22/track49.mp3";
req = WebRequest.Create(testLink) as HttpWebRequest;
resp = req.GetResponse() as HttpWebResponse;
str = resp.GetResponseStream();
int readBytes = 0;
int totalBytes = 0;
byte[] buffer = new byte[1024];
byte[] mp3Buffer = new byte[45000];
IMp3FrameDecompressor decompressor = null;
BufferedWaveProvider provider = null;
WaveOut player = new WaveOut();
Mp3Frame frame = Mp3Frame.LoadFromStream(new ReadFullyStream(str));
while ((readBytes = str.Read(buffer, 0, buffer.Length)) > 0)
{
totalBytes += readBytes;
if (decompressor == null && totalBytes > 1024)
{
Mp3WaveFormat format = new Mp3WaveFormat(frame.SampleRate,
frame.ChannelMode == ChannelMode.Mono ? 1 : 2,
frame.FrameLength, frame.BitRate);
decompressor = new AcmMp3FrameDecompressor(format);
provider = new BufferedWaveProvider(format);
int x = decompressor.DecompressFrame(frame, mp3Buffer, 0);
provider.AddSamples(mp3Buffer, 0, x);
player.Init(provider);
player.Play();
continue;
}
Application.DoEvents();
if (decompressor == null)
continue;
int decNum = decompressor.DecompressFrame(frame, mp3Buffer, 0);
provider.AddSamples(mp3Buffer, 0, decNum);
}
the WaveFormat of your BufferedWaveProvider should be the PCM format you decompress the MP3 into, not an Mp3WaveFormat. Your WaveOut device can only play PCM.
Related
I'm trying to write an app that will listen to my computer audio and transcribe it using Google Speach Recognition.
I've been able to record the system sound using WasapiLoopbackCapture and I've been able to use google streaming recognition api with test files, but I was not able to merge the two togther.
When I stream the audio from the WasapiLoopbackCapture to google it doesn't return any result.
I've based my code on the google code sample at:
https://github.com/GoogleCloudPlatform/dotnet-docs-samples/blob/9588cee6d96bfe484c8e189e9ac2f6eaa3c3b002/speech/api/Recognize/InfiniteStreaming.cs#L225
private WaveInEvent StartListening()
{
var waveIn = new WaveInEvent
{
DeviceNumber = 0,
WaveFormat = new WaveFormat(SampleRate, ChannelCount)
};
waveIn.DataAvailable += (sender, args) =>
_microphoneBuffer.Add(ByteString.CopyFrom(args.Buffer, 0, args.BytesRecorded));
waveIn.StartRecording();
return waveIn;
}
And adjusted it to use the WasapiLoopbackCapture:
private IDisposable StartListening()
{
var waveIn = new WasapiLoopbackCapture();
//var waveIn = new WaveInEvent
//{
// DeviceNumber = 0,
// WaveFormat = new WaveFormat(SampleRate, ChannelCount)
//};
SampleRate = waveIn.WaveFormat.SampleRate;
ChannelCount = waveIn.WaveFormat.Channels;
BytesPerSecond = SampleRate * ChannelCount * BytesPerSample;
Console.WriteLine(SampleRate);
Console.WriteLine(BytesPerSecond);
waveIn.DataAvailable += (sender, args) =>
_microphoneBuffer.Add(ByteString.CopyFrom(args.Buffer, 0, args.BytesRecorded));
waveIn.StartRecording();
return waveIn;
}
But it doesn't return any transcribed text.
I've saved the input stream to a file, and it played ok - so the sound is getting there, my guess is that the waveFormat that is received from the WasapiLoopback is not compatible with what google likes - I tried some conversion and couldn't get it to work.
I've reviewed the following topics on stack overflow, but still couldn't get it to work:
Resampling WasapiLoopbackCapture
Naudio - Convert 32 bit wav to 16 bit wav
And tried combining them both:
private IDisposable StartListening()
{
var waveIn = new WasapiLoopbackCapture();
//var waveIn = new WaveInEvent
//{
//DeviceNumber = 0,
//WaveFormat = new WaveFormat(SampleRate, ChannelCount)
//};
// SampleRate = waveIn.WaveFormat.SampleRate;
// ChannelCount = waveIn.WaveFormat.Channels;
// BytesPerSecond = waveIn.WaveFormat.AverageBytesPerSecond;// SampleRate * ChannelCount * BytesPerSample;
var target = new WaveFormat(SampleRate, 16, 1);
var writer = new WaveFileWriter(#"c:\temp\xx.wav", waveIn.WaveFormat);
Console.WriteLine(SampleRate);
Console.WriteLine(BytesPerSecond);
var stop = false;
waveIn.DataAvailable += (sender, args) =>
{
var a = args;
byte[] newArray16Bit = new byte[args.BytesRecorded / 2];
short two;
float value;
for (int i = 0, j = 0; i < args.BytesRecorded; i += 4, j += 2)
{
value = (BitConverter.ToSingle(args.Buffer, i));
two = (short)(value * short.MaxValue);
newArray16Bit[j] = (byte)(two & 0xFF);
newArray16Bit[j + 1] = (byte)((two >> 8) & 0xFF);
}
var resampleStream = new NAudio.Wave.Compression.AcmStream(new WaveFormat(waveIn.WaveFormat.SampleRate
,16,waveIn.WaveFormat.Channels), target);
Buffer.BlockCopy(newArray16Bit, 0, resampleStream.SourceBuffer, 0, a.BytesRecorded/2);
int sourceBytesConverted = 0;
var bytes = resampleStream.Convert(a.BytesRecorded/2, out sourceBytesConverted);
var converted = new byte[bytes];
Buffer.BlockCopy(resampleStream.DestBuffer, 9, converted,0, bytes);
a = new WaveInEventArgs(converted,bytes);
_microphoneBuffer.Add(ByteString.CopyFrom(a.Buffer, 0, a.BytesRecorded));
if (writer != null)
{
writer.Write(a.Buffer, 0, a.BytesRecorded);
if (writer.Position > waveIn.WaveFormat.AverageBytesPerSecond * 5)
{
stop = true;
writer.Dispose();
writer = null;
Console.WriteLine("Saved file");
}
}
};
waveIn.StartRecording();
return waveIn;
}
But it doesn't work.
I'm not sure if this is the right path.
A code sample of a fix would be highly appreciated
I tried converting the bit rate etc.. but couldn't get this to work.
I followed this NAudio Demo modified to play ShoutCast.
In my full code I have to resample the incoming audio and stream it again over the network to a network player. Since I get many "clicks and pops", I came back to the demo code and I found that these artifacts are originated after the decoding block.
If I save the incoming stream in mp3 format, it is pretty clear.
When I save the raw decoded data (without other processing than the decoder) I get many audio artifacts.
I wonder whether I am doing some error, even if my code is almost equal to the NAudio demo.
Here the function from the example as modified by me to save the raw data. It is called as a new Thread.
private void StreamMP3(object state)
{
//Configuration config = ConfigurationManager.OpenExeConfiguration(ConfigurationUserLevel.None);
//SettingsSection section = (SettingsSection)config.GetSection("system.net/settings");
this.fullyDownloaded = false;
string url = "http://icestreaming.rai.it/5.mp3";//(string)state;
webRequest = (HttpWebRequest)WebRequest.Create(url);
int metaInt = 0; // blocksize of mp3 data
int framesize = 0;
webRequest.Headers.Clear();
webRequest.Headers.Add("GET", "/ HTTP/1.0");
// needed to receive metadata informations
webRequest.Headers.Add("Icy-MetaData", "1");
webRequest.UserAgent = "WinampMPEG/5.09";
HttpWebResponse resp = null;
try
{
resp = (HttpWebResponse)webRequest.GetResponse();
}
catch (WebException e)
{
if (e.Status != WebExceptionStatus.RequestCanceled)
{
ShowError(e.Message);
}
return;
}
byte[] buffer = new byte[16384 * 4]; // needs to be big enough to hold a decompressed frame
try
{
// read blocksize to find metadata block
metaInt = Convert.ToInt32(resp.GetResponseHeader("icy-metaint"));
}
catch
{
}
IMp3FrameDecompressor decompressor = null;
byteOut = createNewFile(destPath, "salva", "raw");
try
{
using (var responseStream = resp.GetResponseStream())
{
var readFullyStream = new ReadFullyStream(responseStream);
readFullyStream.metaInt = metaInt;
do
{
if (mybufferedWaveProvider != null && mybufferedWaveProvider.BufferLength - mybufferedWaveProvider.BufferedBytes < mybufferedWaveProvider.WaveFormat.AverageBytesPerSecond / 4)
{
Debug.WriteLine("Buffer getting full, taking a break");
Thread.Sleep(500);
}
else
{
Mp3Frame frame = null;
try
{
frame = Mp3Frame.LoadFromStream(readFullyStream, true);
if (metaInt > 0)
UpdateSongName(readFullyStream.SongName);
else
UpdateSongName("No Song Info in Stream...");
}
catch (EndOfStreamException)
{
this.fullyDownloaded = true;
// reached the end of the MP3 file / stream
break;
}
catch (WebException)
{
// probably we have aborted download from the GUI thread
break;
}
if (decompressor == null)
{
// don't think these details matter too much - just help ACM select the right codec
// however, the buffered provider doesn't know what sample rate it is working at
// until we have a frame
WaveFormat waveFormat = new Mp3WaveFormat(frame.SampleRate, frame.ChannelMode == ChannelMode.Mono ? 1 : 2, frame.FrameLength, frame.BitRate);
decompressor = new AcmMp3FrameDecompressor(waveFormat);
this.mybufferedWaveProvider = new BufferedWaveProvider(decompressor.OutputFormat);
this.mybufferedWaveProvider.BufferDuration = TimeSpan.FromSeconds(200); // allow us to get well ahead of ourselves
framesize = (decompressor.OutputFormat.Channels * decompressor.OutputFormat.SampleRate * (decompressor.OutputFormat.BitsPerSample / 8) * 20) / 1000;
//this.bufferedWaveProvider.BufferedDuration = 250;
}
int decompressed = decompressor.DecompressFrame(frame, buffer, 0);
//Debug.WriteLine(String.Format("Decompressed a frame {0}", decompressed));
mybufferedWaveProvider.AddSamples(buffer, 0, decompressed);
while (mybufferedWaveProvider.BufferedDuration.Milliseconds >= 20)
{
byte[] read = new byte[framesize];
mybufferedWaveProvider.Read(read, 0, framesize);
byteOut.Write(read, 0, framesize);
}
}
} while (playbackState != StreamingPlaybackState.Stopped);
Debug.WriteLine("Exiting");
// was doing this in a finally block, but for some reason
// we are hanging on response stream .Dispose so never get there
decompressor.Dispose();
}
}
finally
{
if (decompressor != null)
{
decompressor.Dispose();
}
}
}
OK i found the problem. I included the shoutcast metadata to the MP3Frame.
See the comment "HERE I COLLECT THE BYTES OF THE MP3 FRAME" to locate the correct point to get the MP3 frame with no streaming metadata.
The following code runs without audio artifacts:
private void SHOUTcastReceiverThread()
{
//-*- String server = "http://216.235.80.18:8285/stream";
//String serverPath = "/";
//String destPath = "C:\\temp\\"; // destination path for saved songs
HttpWebRequest request = null; // web request
HttpWebResponse response = null; // web response
int metaInt = 0; // blocksize of mp3 data
int count = 0; // byte counter
int metadataLength = 0; // length of metadata header
string metadataHeader = ""; // metadata header that contains the actual songtitle
string oldMetadataHeader = null; // previous metadata header, to compare with new header and find next song
//CircularQueueStream framestream = new CircularQueueStream(2048);
QueueStream framestream = new QueueStream();
framestream.Position = 0;
bool bNewSong = false;
byte[] buffer = new byte[512]; // receive buffer
byte[] dec_buffer = new byte[decSIZE];
Mp3Frame frame;
IMp3FrameDecompressor decompressor = null;
Stream socketStream = null; // input stream on the web request
// create web request
request = (HttpWebRequest)WebRequest.Create(server);
// clear old request header and build own header to receive ICY-metadata
request.Headers.Clear();
request.Headers.Add("GET", serverPath + " HTTP/1.0");
request.Headers.Add("Icy-MetaData", "1"); // needed to receive metadata informations
request.UserAgent = "WinampMPEG/5.09";
// execute request
try
{
response = (HttpWebResponse)request.GetResponse();
}
catch (Exception ex)
{
Console.WriteLine(ex.Message);
return;
}
// read blocksize to find metadata header
metaInt = Convert.ToInt32(response.GetResponseHeader("icy-metaint"));
try
{
// open stream on response
socketStream = response.GetResponseStream();
var readFullyStream = new ReadFullyStream(socketStream);
frame = null;
// rip stream in an endless loop
do
{
if (IsBufferNearlyFull)
{
Debug.WriteLine("Buffer getting full, taking a break");
Thread.Sleep(500);
frame = null;
}
else
{
int bufLen = readFullyStream.Read(buffer, 0, buffer.Length);
try
{
if (framestream.CanRead && framestream.Length > 512)
frame = Mp3Frame.LoadFromStream(framestream);
else
frame = null;
}
catch (Exception ex)
{
frame = null;
}
if (bufLen < 0)
{
Debug.WriteLine("Buffer error 1: exit.");
return;
}
// processing RAW data
for (int i = 0; i < bufLen; i++)
{
// if there is a header, the 'headerLength' would be set to a value != 0. Then we save the header to a string
if (metadataLength != 0)
{
metadataHeader += Convert.ToChar(buffer[i]);
metadataLength--;
if (metadataLength == 0) // all metadata informations were written to the 'metadataHeader' string
{
string fileName = "";
string fileNameRaw = "";
// if songtitle changes, create a new file
if (!metadataHeader.Equals(oldMetadataHeader))
{
// flush and close old byteOut stream
if (byteOut != null)
{
byteOut.Flush();
byteOut.Close();
byteOut = null;
}
if (byteOutRaw != null)
{
byteOutRaw.Flush();
byteOutRaw.Close();
byteOutRaw = null;
}
timeStart = timeEnd;
// extract songtitle from metadata header. Trim was needed, because some stations don't trim the songtitle
//fileName = Regex.Match(metadataHeader, "(StreamTitle=')(.*)(';StreamUrl)").Groups[2].Value.Trim();
fileName = Regex.Match(metadataHeader, "(StreamTitle=')(.*)(';)").Groups[2].Value.Trim();
// write new songtitle to console for information
if (fileName.Length == 0)
fileName = "shoutcast_test";
fileNameRaw = fileName + "_raw";
framestream.reSetPosition();
SongChanged(this, metadataHeader);
bNewSong = true;
// create new file with the songtitle from header and set a stream on this file
timeEnd = DateTime.Now;
if (bWrite_to_file)
{
byteOut = createNewFile(destPath, fileName, "mp3");
byteOutRaw = createNewFile(destPath, fileNameRaw, "raw");
}
timediff = timeEnd - timeStart;
// save new header to 'oldMetadataHeader' string, to compare if there's a new song starting
oldMetadataHeader = metadataHeader;
}
metadataHeader = "";
}
}
else // write mp3 data to file or extract metadata headerlength
{
if (count++ < metaInt) // write bytes to filestream
{
//HERE I COLLECT THE BYTES OF THE MP3 FRAME
framestream.Write(buffer, i, 1);
}
else // get headerlength from lengthbyte and multiply by 16 to get correct headerlength
{
metadataLength = Convert.ToInt32(buffer[i]) * 16;
count = 0;
}
}
}//for
if (bNewSong)
{
decompressor = createDecompressor(frame);
bNewSong = false;
}
if (frame != null && decompressor != null)
{
framedec(decompressor, frame);
}
// fine Processing dati RAW
}//Buffer is not full
SHOUTcastStatusProcess();
} while (playbackState != StreamingPlaybackState.Stopped);
} //try
catch (Exception ex)
{
Console.WriteLine(ex.Message);
}
finally
{
if (byteOut != null)
byteOut.Close();
if (socketStream != null)
socketStream.Close();
if (decompressor != null)
{
decompressor.Dispose();
decompressor = null;
}
if (null != request)
request.Abort();
if (null != framestream)
framestream.Dispose();
if (null != bufferedWaveProvider)
bufferedWaveProvider.ClearBuffer();
//if (null != bufferedWaveProviderOut)
// bufferedWaveProviderOut.ClearBuffer();
if (null != mono16bitFsinStream)
{
mono16bitFsinStream.Close();
mono16bitFsinStream.Dispose();
}
if (null != middleStream2)
{
middleStream2.Close();
middleStream2.Dispose();
}
if (null != resampler)
resampler.Dispose();
}
}
public class QueueStream : MemoryStream
{
long ReadPosition = 0;
long WritePosition = 0;
public QueueStream() : base() { }
public override int Read(byte[] buffer, int offset, int count)
{
Position = ReadPosition;
var temp = base.Read(buffer, offset, count);
ReadPosition = Position;
return temp;
}
public override void Write(byte[] buffer, int offset, int count)
{
Position = WritePosition;
base.Write(buffer, offset, count);
WritePosition = Position;
}
public void reSetPosition()
{
WritePosition = 0;
ReadPosition = 0;
Position = 0;
}
}
private void framedec(IMp3FrameDecompressor decompressor, Mp3Frame frame)
{
int Ndecoded_samples = 0;
byte[] dec_buffer = new byte[decSIZE];
Ndecoded_samples = decompressor.DecompressFrame(frame, dec_buffer, 0);
bufferedWaveProvider.AddSamples(dec_buffer, 0, Ndecoded_samples);
NBufferedSamples += Ndecoded_samples;
brcnt_in.incSamples(Ndecoded_samples);
if (Ndecoded_samples > decSIZE)
{
Debug.WriteLine(String.Format("Too many samples {0}", Ndecoded_samples));
}
if (byteOut != null)
byteOut.Write(frame.RawData, 0, frame.RawData.Length);
if (byteOutRaw != null) // as long as we don't have a songtitle, we don't open a new file and don't write any bytes
byteOutRaw.Write(dec_buffer, 0, Ndecoded_samples);
frame = null;
}
private IMp3FrameDecompressor createDecompressor(Mp3Frame frame)
{
IMp3FrameDecompressor dec = null;
if (frame != null)
{
// don't think these details matter too much - just help ACM select the right codec
// however, the buffered provider doesn't know what sample rate it is working at
// until we have a frame
WaveFormat srcwaveFormat = new Mp3WaveFormat(frame.SampleRate, frame.ChannelMode == ChannelMode.Mono ? 1 : 2, frame.FrameLength, frame.BitRate);
dec = new AcmMp3FrameDecompressor(srcwaveFormat);
bufferedWaveProvider = new BufferedWaveProvider(dec.OutputFormat);// decompressor.OutputFormat
bufferedWaveProvider.BufferDuration = TimeSpan.FromSeconds(400); // allow us to get well ahead of ourselves
// ------------------------------------------------
//Create an intermediate format with same sampling rate, 16 bit, mono
middlewavformat = new WaveFormat(dec.OutputFormat.SampleRate, 16, 1);
outwavFormat = new WaveFormat(Fs_out, 16, 1);
// wave16ToFloat = new Wave16ToFloatProvider(provider); // I have tried with and without this converter.
wpws = new WaveProviderToWaveStream(bufferedWaveProvider);
//Check middlewavformat.Encoding == WaveFormatEncoding.Pcm;
mono16bitFsinStream = new WaveFormatConversionStream(middlewavformat, wpws);
middleStream2 = new BlockAlignReductionStream(mono16bitFsinStream);
resampler = new MediaFoundationResampler(middleStream2, outwavFormat);
}
return dec;
}
I try to read a .raw audio file from the Android External Storage directory by using FileInputStream. Sometimes, I can read the file, sometimes not. What is the reason for this case?
Any help will be appreciated.
storageDir = Environment.getExternalStorageDirectory().getAbsolutePath();
enrollWavPath = storageDir + 1 + ".raw";
File file = new File(enrollRawPath);
byte[] byteData = null;
int size = (int) file.length();
byteData = new byte[(int)(size)];
FileInputStream iFileStream = new FileInputStream(file);
int bytesread = 0;
int ret = 0;
while (bytesread < size) {
ret = iFileStream.read( byteData,0, 2*count);
if (ret != -1) {
shortData = byte2short(byteData);
bytesread += ret;
}
else {
break;
}
}
iFileStream.close();
I am trying to load image from a folder into my application, the code I am using is
FileConnection fc = null;
DataInputStream in = null;
DataOutputStream out = null;
try {
fc = (FileConnection)Connector.open("file:///e:/Images/Abc.jpg");
int length = (int)fc.fileSize();//possible loss of precision may throw error
byte[] data = null;
if (length != -1) {
data = new byte[length];
in = new DataInputStream(fc.openInputStream());
in.readFully(data);
}
else {
int chunkSize = 112;
int index = 0;
int readLength = 0;
in = new DataInputStream(fc.openInputStream());
data = new byte[chunkSize];
do {
if (data.length < index + chunkSize) {
byte[] newData = new byte[index + chunkSize];
System.arraycopy(data, 0, newData, 0, data.length);
data = newData;
}
readLength = in.read(data, index, chunkSize);
index += readLength;
} while (readLength == chunkSize);
length = index;
}
Image image = Image.createImage(data, 0, length);
image = createThumbnail(image);
ImageItem imageItem = new ImageItem(null, image, 0, null);
mForm.append(imageItem);
mForm.setTitle("Done.");
fc = (FileConnection)Connector.open("file:///e:/Images/Abc.jpg");
if(!fc.exists()){
try{
fc.create();
}catch(Exception ce){System.out.print("Create Error: " + ce);}
}
out = new DataOutputStream(fc.openOutputStream());
out.write(data);
}
catch (IOException ioe) {
StringItem stringItem = new StringItem(null, ioe.toString());
mForm.append(stringItem);
mForm.setTitle("Done.");
}
finally {
try {
if (in != null) in.close();
if (fc != null) fc.close();
}
catch (IOException ioe) {}
hope it can be the problem with image I tried to resize the image
private Image createThumbnail(Image image)
{
int sourceWidth = image.getWidth();
int sourceHeight = image.getHeight();
int thumbWidth = 18;
int thumbHeight = 23;
Image thumb = Image.createImage(thumbWidth, thumbHeight);
Graphics g = thumb.getGraphics();
for (int y = 0; y < thumbHeight; y++)
{
for (int x = 0; x < thumbWidth; x++)
{
g.setClip(x, y, 1, 1);
int dx = x * sourceWidth / thumbWidth;
int dy = y * sourceHeight / thumbHeight;
g.drawImage(image, x - dx, y - dy, Graphics.LEFT | Graphics.TOP);
}
}
Image immutableThumb = Image.createImage(thumb);
return immutableThumb;
}
Still it returns an exception out of memmory native image deocde error, Somebody pls help me to sort it out
This may occur becuase of two reason.
Image size: In some phones due to small heap size not device does not load image so try with smaller image size.
Second reason may be image is corrupt so try with another image.
I'm using Speex to encode the raw data but after I decode the data the audio plays at a faster rate because it makes you sound like a chipmunk. I'm using NSpeex and Silverlight 4.
8kHz Sampling
Encoding Function:
JSpeexEnc encoder = new JSpeexEnc();
int rawDataSize = 0;
public byte[] EncodeAudio(byte[] rawData)
{
var encoder = new SpeexEncoder(BandMode.Narrow);
var inDataSize = rawData.Length / 2;
var inData = new short[inDataSize];
for (var index = 0; index < rawData.Length; index += 2)
{
inData[index / 2] = BitConverter.ToInt16(rawData, index);
}
inDataSize = inDataSize - inDataSize % encoder.FrameSize;
var encodedData = new byte[rawData.Length];
var encodedBytes = encoder.Encode(inData, 0, inDataSize, encodedData, 0, encodedData.Length);
byte[] encodedAudioData = null;
if (encodedBytes != 0)
{
encodedAudioData = new byte[encodedBytes];
Array.Copy(encodedData, 0, encodedAudioData, 0, encodedBytes);
}
rawDataSize = inDataSize; // Count of encoded shorts, for debugging
return encodedAudioData;
}
Decoding Function:
SpeexDecoder decoder = new SpeexDecoder(BandMode.Narrow);
public byte[] Decode(byte[] encodedData)
{
try
{
short[] decodedFrame = new short[8000]; // should be the same number of samples as on the capturing side
int decoderBytes = decoder.Decode(encodedData, 0, encodedData.Length, decodedFrame, 0, false);
byte[] decodedData = new byte[encodedData.Length];
byte[] decodedAudioData = null;
decodedAudioData = new byte[decoderBytes * 2];
for (int shortIndex = 0, byteIndex = 0; byteIndex < decoderBytes; shortIndex++)
{
BitConverter.GetBytes(decodedFrame[shortIndex + byteIndex]).CopyTo(decodedAudioData, byteIndex * 2);
byteIndex++;
}
// todo: do something with the decoded data
return decodedAudioData;
}
catch (Exception ex)
{
ShowMessageBox(ex.Message.ToString());
return null;
}
}
Playing the audio:
void PlayWave(byte[] PCMBytes)
{
byte[] decodedBuffer = Decode(PCMBytes);
MemoryStream ms_PCM = new MemoryStream(decodedBuffer);
MemoryStream ms_Wave = new MemoryStream();
_pcm.SavePcmToWav(ms_PCM, ms_Wave, 16, 8000, 1);
WaveMediaStreamSource WaveStream = new WaveMediaStreamSource(ms_Wave);
mediaElement1.SetSource(WaveStream);
mediaElement1.Play();
}
Sorry guys for the late response but I figured out what the problem was.
Inside my decode function I loop through the decoded short array but I'm only copying half of the bytes into my new byte array.
It needs to look something like this:
decodedAudioData = new byte[decoderBytes * 2];
for (int shortIndex = 0, byteIndex = 0; shortIndex < decodedFrame.Length; shortIndex++, byteIndex += 2)
{
byte[] temp = BitConverter.GetBytes(decodedFrame[shortIndex]);
decodedAudioData[byteIndex] = temp[0];
decodedAudioData[byteIndex + 1] = temp[1];
}