[Openal] How can I play mono wav sound in specific channel?
Ümit Uzun
umituzun84 at gmail.com
Thu Apr 16 06:06:03 PDT 2009
Hi Chris;
I have implemented your code in Framework.cpp and it works like a charm :D
Thanks so much for your awesome helps :D But I have some questions to solve.
I have implemented your suggestion on 5.1 speaker system like that;
template<typename T>
T *load_data(const T *data, int size)
{
size /= sizeof(T);
T *data71 = new T[size*6];
for(int i = 0;i < size;i++) {
data71[i*6 + 0] = 0; // front-left
data71[i*6 + 1] = 0; // front-right
data71[i*6 + 2] = 0; // front-center
data71[i*6 + 3] = 0; // lfe (sub-woofer)
data71[i*6 + 4] = 0; // back-left
data71[i*6 + 5] = 0; // back-right
}
return data71;
}
--------------------------------------
WAVEFORMATEX fmt;
g_pWaveLoader->GetWaveFormatExHeader(WaveID, &fmt);
if(fmt.wBitsPerSample == 8) {
ALubyte *data = load_data((const ALubyte*)pData, iDataSize);
alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_51CHN8"),
data, iDataSize*6, iFrequency);
delete[] data;
} else if(fmt.wBitsPerSample == 16) {
ALshort *data = load_data((const ALshort*)pData, iDataSize);
alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_51CHN16"),
data, iDataSize*6, iFrequency);
delete[] data;
} else if(fmt.wBitsPerSample == 32) { // assumes 32-bit float
ALfloat *data = load_data((const ALfloat*)pData, iDataSize);
alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_51CHN32"),
data, iDataSize*6, iFrequency);
delete[] data;
}
It works on while using Generic Hardware. I mean I can direct each sound to
each wanted channel errorless, but when I try same operation while using
Generic Software it works but it is mixing the channels. For example I want
to play on back- left, but sound is heard from front-left, back-left,
front-center and lfe and vice-versa for the back-right.
I mean Generic Software can't play related channel on related side. Do you
think it is problem of generic software or what? I have looked at the
OpenALProgrammingGuide and there is a statement about kind of this situation
but I can't really understand what does it mean? Can you explain what could
be the problem?
>From OpenALProgrammingGuide;
"The multi-channel extension provides a mechanism to play multi-channel data
via OpenAL. A variety of formats are supported. Multi-channel buffers can be
attached or queued on a source.
Note that when using the “Generic Software” device, the multi-channel
buffers are mixed down to a stereo output. On a hardware device (such as the
“Generic Hardware” device or a native device), each channel of a buffer
requires a hardware voice. So, for example playing a buffer using the
AL_FORMAT_51CHN16 format will require 6 free hardware voices. If the
hardware resources are unavailable, the call to alSourceQueueBuffers or
alSourcei will fail."
Thanks so much.
Best Regards.
2009/4/16 Chris Robinson <chris.kcat at gmail.com>
> On Thursday 16 April 2009 2:53:45 am you wrote:
> > Hi Chris;
> >
> > > ALshort monodata[samples];
> > >
> > > you can fill it into a 7.1 buffer like this:
> > >
> > > ALshort data71[samples*8];
> > > for(int i = 0;i < samples;i++) {
> > > data71[i*8 + 0] = 0; // front-left
> > > data71[i*8 + 1] = 0; // front-right
> > > data71[i*8 + 2] = 0; // front-center
> > > data71[i*8 + 3] = 0; // lfe (sub-woofer)
> > > data71[i*8 + 4] = 0; // back-left
> > > data71[i*8 + 5] = 0; // back-right
> > > data71[i*8 + 6] = monodata[i]; // side-left
> > > data71[i*8 + 7] = monodata[i]; // side-right
> > > }
> > > alBufferData(bID, alGetEnumValue("AL_FORMAT_71CHN16"), data71,
> > > samples*8*sizeof(ALshort), frequency);
> > >
> > I have look at the Framework and there is ALFWLoadWaveToBuffer;
> >
> > ALboolean ALFWLoadWaveToBuffer(const char *szWaveFile, ALuint uiBufferID,
> > ALenum eXRAMBufferMode)
> > {
> > WAVEID WaveID;
> > ALint iDataSize, iFrequency;
> > ALenum eBufferFormat;
> > ALchar *pData;
> > ALboolean bReturn;
> >
> > bReturn = AL_FALSE;
> > if(g_pWaveLoader)
> > {
> > if(SUCCEEDED(g_pWaveLoader->LoadWaveFile(szWaveFile, &WaveID)))
> > {
> > if((SUCCEEDED(g_pWaveLoader->GetWaveSize(WaveID, (unsigned
> > long*)&iDataSize))) &&
> > (SUCCEEDED(g_pWaveLoader->GetWaveData(WaveID,
> > (void**)&pData))) &&
> > (SUCCEEDED(g_pWaveLoader->GetWaveFrequency(WaveID,
> (unsigned
> > long*)&iFrequency))) &&
> > (SUCCEEDED(g_pWaveLoader->GetWaveALBufferFormat(WaveID,
> > &alGetEnumValue, (unsigned long*)&eBufferFormat))))
> > {
> > // Set XRAM Mode (ifapplication)
> > if(eaxSetBufferMode && eXRAMBufferMode)
> > {
> > eaxSetBufferMode(1, &uiBufferID, eXRAMBufferMode);
> > }
> >
> > alGetError();
> > alBufferData(uiBufferID, eBufferFormat, pData, iDataSize,
> > iFrequency);
> >
> > if(alGetError() == AL_NO_ERROR)
> > {
> > bReturn = AL_TRUE;
> > }
> >
> > g_pWaveLoader->DeleteWaveFile(WaveID);
> > }
> > }
> > }
> >
> > return bReturn;
> > }
> >
> > You have given ALshort monodata[samples]; but in this code ALchar *pData;
> > This variable are same or not. because alBufferData takes void format and
> > ALchar is 1 byte and ALshort 2 byte. I can't get what should I do.
>
> They aren't the same. That function uses pData because the loaded wave data
> may be 8-bit, or 16-bit (or possible 32-bit), so it just uses a generic
> char
> type.
>
> g_pWaveLoader has a GetWaveFormatExHeader method you can use to get a
> WAVEFORMATEX struct, which has a wBitsPerSample field. You need to check
> that
> field and act on the data depending on the value (eg. if it's 8, use
> ALubyte,
> if it's 16, use ALshort).
>
> And example of doing this in C++ would be something like:
>
> template<typename T>
> T *load_data(const T *data, int size)
> {
> size /= sizeof(T);
>
> T *data71 = new T[size*8];
> for(int i = 0;i < size;i++) {
> data71[i*8 + 0] = 0; // front-left
> data71[i*8 + 1] = 0; // front-right
> data71[i*8 + 2] = 0; // front-center
> data71[i*8 + 3] = 0; // lfe (sub-woofer)
> data71[i*8 + 4] = 0; // back-left
> data71[i*8 + 5] = 0; // back-right
> data71[i*8 + 6] = data[i]; // side-left
> data71[i*8 + 7] = data[i]; // side-right
> }
>
> return data71;
> }
>
> ...
>
> WAVEFORMATEX fmt;
>
> g_pWaveLoader->GetWaveFormatExHeader(WaveID, &fmt);
> if(fmt.wBitsPerSample == 8) {
> ALubyte *data = load_data((const ALubyte*)pData, iDataSize);
> alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_71CHN8"),
> data, iDataSize*8, iFrequency);
> delete[] data;
> } else if(fmt.wBitsPerSample == 16) {
> ALshort *data = load_data((const ALshort*)pData, iDataSize);
> alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_71CHN16"),
> data, iDataSize*8, iFrequency);
> delete[] data;
> } else if(fmt.wBitsPerSample == 32) { // assumes 32-bit float
> ALshort *data = load_data((const ALfloat*)pData, iDataSize);
> alBufferData(uiBufferID, alGetEnumValue("AL_FORMAT_71CHN32"),
> data, iDataSize*8, iFrequency);
> delete[] data;
> }
>
> ..
> completely untested, but the concept should work. It would load the sound
> so
> that when it plays, it'll play out the side speakers.
> _______________________________________________
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> Openal at opensource.creative.com
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>
--
Ümit Uzun
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