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Samples normalization

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	Samples normalization by Fruskus :: Rate this Message: Reply to Author \| View Threaded \| Show Only this Message Hi, this is the first time I post here. First of all, I'd like to thank all of you, because this forum has been really helpful to me. I'm doing a real time filtering application. So I'm using the capturing and playing with queue routine. But when I modify the buffer, values may exceed the short max values. Does anyone know how to normalizate buffers in realtime? I've tried to use a normalization loop, but I think this only work for a whole file, not for streaming. Thanks in advance.
	Re: Samples normalization by Daniel PEACOCK :: Rate this Message: Reply to Author \| View Threaded \| Show Only this Message Hi, When ever you are processing audio in a way that means there is a chance for a sample to exceed the range of signed short, then you need to temporarily convert the samples into some other format (e.g signed longs or floats). After processing is completed, you should clamp the values into the allowable range (-32768 to 32767) and convert the samples back to signed shorts. This may introduce clipping, but that is better than the alternative of overflows where the most significant bits of the magnitude are lost. Dan Creative Labs (UK) Ltd. Notice The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying or distribution of the message, or any action taken by you in reliance on it, is prohibited and may be unlawful. If you have received this message in error, please delete it and contact the sender immediately. Thank you. Creative Labs UK Ltd company number 2658256 registered in England and Wales at Belmont Road, Belmont Place, Maidenhead, Berkshire, SL6 6TB Fruskus <karlonchiz@hotma il.com> To Sent by: openal@... openal-bounces@op cc ensource.creative .com Subject [Openal] Samples normalization 07/03/2009 01:03 PM Hi, this is the first time I post here. First of all, I'd like to thank all of you, because this forum has been really helpful to me. I'm doing a real time filtering application. So I'm using the capturing and playing with queue routine. But when I modify the buffer, values may exceed the short max values. Does anyone know how to normalizate buffers in realtime? I've tried to use a normalization loop, but I think this only work for a whole file, not for streaming. Thanks in advance. -- View this message in context: http://www.nabble.com/Samples-normalization-tp24322038p24322038.html Sent from the OpenAL - User mailing list archive at Nabble.com. _______________________________________________ Openal mailing list Openal@... http://opensource.creative.com/mailman/listinfo/openal ForwardSourceID:NT0006DBEE _______________________________________________ Openal mailing list Openal@... http://opensource.creative.com/mailman/listinfo/openal
	Re: Samples normalization by Fruskus :: Rate this Message: Reply to Author \| View Threaded \| Show Only this Message Daniel PEACOCK wrote: Hi, When ever you are processing audio in a way that means there is a chance for a sample to exceed the range of signed short, then you need to temporarily convert the samples into some other format (e.g signed longs or floats). After processing is completed, you should clamp the values into the allowable range (-32768 to 32767) and convert the samples back to signed shorts. This may introduce clipping, but that is better than the alternative of overflows where the most significant bits of the magnitude are lost. Dan Creative Labs (UK) Ltd. Fruskus Hi, this is the first time I post here. First of all, I'd like to thank all of you, because this forum has been really helpful to me. I'm doing a real time filtering application. So I'm using the capturing and playing with queue routine. But when I modify the buffer, values may exceed the short max values. Does anyone know how to normalizate buffers in realtime? I've tried to use a normalization loop, but I think this only work for a whole file, not for streaming. Thanks in advance. Thanks Dan, that was what I was doing, and it works. But I was just wondering If there was a better way to do this, and avoid the variations between the differents buffers signal level.
	Re: Samples normalization by Daniel PEACOCK :: Rate this Message: Reply to Author \| View Threaded \| Show Only this Message Hi, So are you concerned about the range in volume of the samples contained in a single processed buffer, or how loud one buffer is compared to another buffer? Dan Creative Labs (UK) Ltd. Notice The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying or distribution of the message, or any action taken by you in reliance on it, is prohibited and may be unlawful. If you have received this message in error, please delete it and contact the sender immediately. Thank you. Creative Labs UK Ltd company number 2658256 registered in England and Wales at Belmont Road, Belmont Place, Maidenhead, Berkshire, SL6 6TB Fruskus <karlonchiz@hotma il.com> To Sent by: openal@... openal-bounces@op cc ensource.creative .com Subject Re: [Openal] Samples normalization 07/03/2009 02:44 PM Daniel PEACOCK wrote: > > > Hi, > > When ever you are processing audio in a way that means there is a chance > for a sample to exceed the range of signed short, then you need to > temporarily convert the samples into some other format (e.g signed longs > or > floats). After processing is completed, you should clamp the values into > the allowable range (-32768 to 32767) and convert the samples back to > signed shorts. This may introduce clipping, but that is better than the > alternative of overflows where the most significant bits of the magnitude > are lost. > > Dan > Creative Labs (UK) Ltd. > > Fruskus > > > > > > > > Hi, this is the first time I post here. > > First of all, I'd like to thank all of you, because this forum has been > really helpful to me. > > I'm doing a real time filtering application. So I'm using the capturing > and > playing with queue routine. But when I modify the buffer, values may > exceed > the short max values. > > Does anyone know how to normalizate buffers in realtime? I've tried to use > a > normalization loop, but I think this only work for a whole file, not for > streaming. > > Thanks in advance. > > > Thanks Dan, that was what I was doing, and it works. But I was just wondering If there was a better way to do this, and avoid the variations between the differents buffers signal level. -- View this message in context: http://www.nabble.com/Samples-normalization-tp24322038p24323376.html Sent from the OpenAL - User mailing list archive at Nabble.com. _______________________________________________ Openal mailing list Openal@... http://opensource.creative.com/mailman/listinfo/openal ForwardSourceID:NT0006DBFE _______________________________________________ Openal mailing list Openal@... http://opensource.creative.com/mailman/listinfo/openal
	Re: Samples normalization by Fruskus :: Rate this Message: Reply to Author \| View Threaded \| Show Only this Message Daniel PEACOCK wrote: Hi, So are you concerned about the range in volume of the samples contained in a single processed buffer, or how loud one buffer is compared to another buffer? That was my point. But anyway, I've run some tests and it's working ok and the differences aren't noticeable. Let's see how it works when I start using filters.