NAudio 1.3 Release Notes

I guess there is a problem in dsp/BiQuadFilter.cs Transform():
for(int n = 0; n < inBuffer.Length; n++)
{
y[n] = (float) (
(b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
- (a1/a0)*y[n-1] - (a2/a0)*y[n-2]);
}

Index out of boudaries.

Thanks for replay,
I need simple low latency spectrum visualization just for 3-6 bands. So I guess several BiQuadFilters is the best choice.
It`s sad that I failed utilise this nice nubble from NAudio.

Seems it works since initial n value I changed from 0 to 2. Low pass filter seems to be working, maybe later I`ll check outputs quality.

Congratulations with this new release!

Hi Mark,
Today I tried FFT function from NAudio, works fine. Seems simplier resort for my task than using BiQuadFilter.

I have another problm with selecting input device. WaveIn.DeviceCount shows only one input device however in windows Record Control panel i can pick from several(Mic, line-in etc). Maybe is any resort from selecting from own application? (running on XP)

One more obscurity about waveIn recording channels, how it works in mono mode, record only left or right or averages samples from both?

Hey Mark,
Looks like some amazing work you've got into NAudio.

I am looking at your re-sampler project example and wondering if there is a simple way to use a similar set of commands to loop through a wave file's samples and tweak the volume before writing out to a new file. I would like to not just reset the volume level on the entire file, but have a minute control over the volume at any given moment in the audio so I can fade in and out throughout the file and whatnot.
Thanks for any input!
Jason

Hi Mark, cool to see that NAudio is constantly being improved!

Is there anyway of converting wma file into a MuLaw wav format

I noticed the IAudioSessionManager2 is not yet implemented in your code.

Can you please advise me what tools you are using to get method signatures generated?

Mark,
NAudio looks pretty cool. I'm looking for a replacement for DirectSound now that MS has deprecated it. My app uses most of the DirectSound API. Is there an API manual or programmers guide for NAudio? I didn't see a manual on CodePlex. Thanks.

Ok, thanks Mark. I'll check out the source code.

Can i use your library to play an online radio streaming in MP3/WMA/OGG formats?

Hi Mark,

Reading the release notes, I came accross this:

"Mp3FileReader and WaveFileReader can have a stream as input, and WaveFileWriter can write to a stream. These features are useful to those wanting to stream data over networks."

Are there any examples of how to have WaveFileWriter write to a stream? This would make NAudio the missing link in my project. Any help here would be greatly appreciated.

Mark,

Just want to say a big thanks for all your hard work on NAudio, have found it an incredibly useful tool. One question - now that WaveInStream is obsolete in favour of WaveIn, is it possible to change the default input device? Is there a WaveIn equivalent to WaveInStream(int deviceNumber ... ...) where deviceNumber is the mixer I wish to use?

Just to be clear, I am not asking how to change between e.g. Stereo Mix and Microphone sources (know how to do that via the Mixer API). But I have multiple WaveIn devices and want to choose between them.

Thanks in advance

CD

So simple, thanks.

One feature I have been toying with is the use of loopback capture (basically recording directly from the speakers, thus getting round soundcards that don't have e.g. Stereo Mix or the like). Have had some success thanks to a couple of links on Codeplex (most notably
http://naudio.codeplex.com/Thread/View.aspx?ThreadId=75005)
. However, this solution seems limited in C# because the GC seems to be unable to cope if there is any silence at all in the audio (basically it does record audio, but the recorded clip skips and jumps all the time). I saw a suggestion here of how to fix it -

http://blogs.msdn.com/matthew_van_eerde/archive/2008/12/10/sample-playing-silence-via-wasapi-event-driven-pull-mode.aspx

but my C++ interop skills are nowhere near good enough. Wondered if you had any other ideas of how to solve this? Would love to see this feature in the future if you are looking for ways to improve an already excellent library.

Thanks once again

CD

hi mark,

first of all thanks for this wonderful library, its really useful. secondly im currently working on my final year project and i need a real time live audio input so that i can save it to an array. how is the best way to implement this using the NAudio library?

thanks tristan

thanks for the quick replay

i managed to do the first tutorial on sound recording and saving the file to .wav, know i dont want to save the recorded file as .wav, but i want to save the data to an array. know im using this part of the code:

void waveIn_DataAvailable(object sender, WaveInEventArgs e)
{
byte[] buffer = e.Buffer;
int bytesRecorded = e.BytesRecorded;
WriteToFile(buffer, bytesRecorded);

will the recorded audio be sent to array e.buffer?

thanks tristan

hi mark,
i just need to create a button which allows me to record from a line in from the pc sound card and save the contains into a buffer...any help? i already tried but im a newbie and i got no idea what im doing

thanks tristan

Hi Mark,

Excellent project, professionally done.

NAudio 1.3 has problems playing an MP3 file where the MP3Header does not immediately follow the ID3Tags.

The MP3Header must be scanned after all known headers/tags have been found.
// try for a header
long headerStartPosition = input.Position;
byte[] headerBytes = reader.ReadBytes(4);

// Added -jam to play wrapped mp3 files via RIFF
headerBytes = CheckForRiff(input, reader, headerBytes);

bool bHeaderFound = false;

long readsToTry = 4000;
if (readsToTry > input.Length)
readsToTry = input.Length;

while (bHeaderFound == false && readsToTry > 0)
{

if ((headerBytes[0] == 0xFF) && ((headerBytes[1] & 0xE0) == 0xE0))
{
bHeaderFound = DecodeMpegHeader(headerBytes);
}

if(bHeaderFound == false)
{
--readsToTry;
headerBytes[0] = headerBytes[1];
headerBytes[1] = headerBytes[2];
headerBytes[2] = headerBytes[3];
headerBytes[3] = reader.ReadByte();
}
}

if (!bHeaderFound)
{
input.Position = headerStartPosition;
throw new FormatException("Not a recognised MP3 block");
}

Ron Nelson

Hi Mark, firstly great library! I'm planning to use it in a 'batch mode' to create individual amplitude sample sets for a large mp3 library. I notice that your demo MeteringStream class does exactly what I need, but in realtime as it intercepts the stream en route to the WavePlayer. Do you have any example code showing simple consumption of the WaveStream with creation of volume samples - I'm getting confused by the multi channel aspects and index += 4; nature of the stream's low level format.

Again, thanks for a great library!

regards

SeanR

Hi Mark - Tom, Sarah, Dan and Jennie here. We think that the key issue is the low latency spectrum visualisation algorithm. Really what you want to do is improve your C++ interop skills to be honest. See you in the morning!

Great job, but I have a question. I searched all your blog and NAudio Discussion threads but found nothing.
I have a GSM file (recorded by asterisk), how can I play it with NAudio?

Hi Mark,
just like CD mentioned above,
I'd like to be able to use loopback-capturing to capture any sound output from the soundcard.

Lots of machines / drivers do not support "Record from stereomix",
so loopback is the only option in those cases.

What would be the correct way to do loopback recording with NAudio ?

Hi!

First of all insanely great stuff!!
Can you tell me how could i default to 89db volume all mp3 that i play ?

Regards,
Alen

Hi Mark, Can you make YouTube Videos demonstrating how to youse NAudio?
Can you upgrade the code to use all the new features of .Net 4.0? C# 4.0?

how can I use Speex Codec With NAudio Over the internet?

So I will record audio from mic then send it to player over the internet?

can you explain NAudio internals and how stuff works in it? for beginners?

What is a wave file, and what does it consist of, etc.. and how to does NAudio work? If you can make an introduction to NAudio in YouTube it will be good. with demos.

I have developed a sound capturing winform application using NAudio 1.3
on c#. The idea is to capture streaming songs through sound card. It is
working perfectly fine on windows xp.
But On vista and windows7 machine if there is no sound driver installed it also take mic voice with it
which is a big issue(i have checked it on dell laptop and if i mute the microphone than it does not capture any sound
)i just want to capture streaming song sound .NAudio 1.3 uses WASAPI audio capture for windows vista and 7.I need to solve this
issue urgently.

Hi Mark,
I have downloaded the Naudio.dll (1.3). but when i try to play .mp3 file it gives "Not a recognised mp3 Block.". Please Provide me the new version of Naudio.dll so i can play mp3 files.

Thanks
Umesh(umesh.prajapati@tatvasoft.com)

Hi Mark, your work is really amazing! I would like to ask you a question ... Is there any possibility to translate the source code to vb.net 2008? I have no knowledge they in c # and I would work together to debug and optimize code, and collaborate with the project. I'm working on a radio automation software. Alejandro (alenet444@hotmail.com)

Hi Mark,
I have downloaded the Naudio.dll (1.3). but when i try to play .mp3 file it gives "Not a recognised mp3 Block.". Please Provide me the new version of Naudio.dll so i can play mp3 files.

Thanks
Umesh(umesh.prajapati@tatvasoft.com)

I am trying to get the pitch of a sound using the fft function. I created an array with the intensity of each frequency and I can get the index of the pitch. Now my problem is. How can I convert this index to a frequency.

Hi Mark, I'm developing a voip application and would like to use NAudio with Wasapi for low latency. However I need to do sample rate conversion. Do you have any recommendations? Can I use the resampler in Speex for real time conversion?

hi I just need a simple code to use naudio so that I can play a music and then by the track bar I can change the volume

does anyone know a way to detect for example the highest 10 peaks? I have implemented FFT Radix2.
I am using an amplitude threshold but in most cases the amplitude threshold is too high and it doesn't detect all peaks.From what I see in excel, after recording peaks, they decrease in time.
Does anyone know another way to detect the highest 10, 11 peaks( or peaks)?

Need help...PLSS

I downloaded your library from the official site. http://naudio.codeplex.com/releases/view/34249

But I couldn`t find any BufferedWaveProvider. This is not even in the source code. Can you check please?