Low Level Audio API question

Does anyone know how to alter the sample rate of a wave once it has been opened with WaveOutOpen? I can alter it if I change the nsamplespersec entry in my WAVEFORMATEX structure but can I alter it once I have opened the wave? Where is this info stored. Anyone have any better ideas on how to speed up and slow down a wave by any other means?

This one is hard guys!!

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Have you looked at MMTools from swiftsoft www.swiftsoft.de. Their wave components do almost anything that can be done with wave audio. There is a pitch component in the audio pack.
I found this:


If it is a complete description of this particular API I would say the answer is you can't :-(


hi dave,

maybe you can also find something at simonets athenas place


there is also a howtoproj with wave-files

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davegreenAuthor Commented:
Adjusted points to 1000
davegreenAuthor Commented:
Thanks guys.. Raymond.. That looks like a api for their own application..not the Win API.
meikl.. already checked this out.. he is not using the low level wave api

Upped the points... I REALLY need an answer for this one!

Is DirectSound (DirectX) an option?
davegreenAuthor Commented:
Adjusted points to 1500
davegreenAuthor Commented:
I didn't want to use DirectSound, although I know that it has a pitch function that would work.
davegreenAuthor Commented:
1500 point should be enough I think for an answer :O)
Have you tried the waveOutSetPlaybackRate function to speed up or slow down your playback?

davegreenAuthor Commented:
I'm not sure this is available on all soundcards, if at all. I tried it on mine and it didn't work. I need something that will work on every wave  playing capable computer.
From the MSDN:
Changing Pitch and Playback Rate
Some waveform-audio output devices can vary the pitch and the playback rate of waveform-audio data. Not all waveform-audio devices support pitch and playback-rate changes. To determine whether a device supports playback rate changes, use the WAVECAPS_PLAYBACKRATE flag to test the dwSupport member of the WAVEOUTCAPS structure (filled by the waveOutGetDevCaps function).

The differences between changing pitch and playback rate are as follows:

Changing the playback rate is performed by the device driver and does not require specialized hardware. The sample rate is not changed, but the driver interpolates by skipping or synthesizing samples. For example, if the playback rate is changed by a factor of two, the driver skips every other sample.
Changing the pitch requires specialized hardware. The playback rate and sample rate are not changed.
Windows provides the following functions to query and set waveform-audio pitch and playback rates.

waveOutGetPitch Retrieves the pitch for the specified waveform-audio output device.

waveOutGetPlaybackRate Retrieves the playback rate for the specified waveform-audio output device.

waveOutSetPitch Sets the pitch for the specified waveform-audio output device.

waveOutSetPlaybackRate Sets the playback rate for the specified waveform-audio output device.

The pitch and playback rates are changed by a factor specified with a fixed-point number packed into a doubleword value. The upper 16 bits specify the integer part of the number; the lower 16 bits specify the fractional part. For example, the value 1.5 is represented as 0x00018000L. The value 0.75 is represented as 0x0000C000L. A value of 1.0 (0x00010000) means the pitch or playback rate is unchanged.

So, if your sound card is not working, then determine what the cause is from the returned error message. It looks like these are the only methods that the windows API supports. Unfortunately we live in an imperfect world and hardware does not always support what we require. Just look at printers to see an example..

Anyway, that's all I can help with unfortunately..
hi dave!

you can do whatever you want with that .WAV file while you opened it as normal file (without API), so:

- open .WAV file as usual AssignFile( F, 'mywav.wav' );
- read it all (or by chunks) to memory
- analyze .WAV header to find out sampling rate, 8 or 16 bit etc...
- now CONVERT it by yourself to another sampling rate... this is called stretching or squeezing algorythm. You better to use Brezenhaim (spelling is wrong, I know ;) approach to get it changed to another frequence. Easy example: if you need it faster two times, just throw out every second byte (or word if 16bit)... But, to make it 'good' conversion you have to apply some dithering.
- now add new .WAV header to that converted data and send it it multimedia API

this is the only way I think....
davegreenAuthor Commented:
These look great! although a little expensive to buy the source. The pitch component looks like it will do the trick.
Thanks again guys!
I need your help strongly......realy strongly....please if you have some tiny spare time just send me an e-mail with a single word "ok".

I am working on a project on the low level wave recording, and I am trying to use the wave\in, wave\out functions....

just a little help from you will lead me out.....

yours daniel mondi.....

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