>>> /* Update phase, rollover at 1.0 */
>>> data->phase[0] += (data->frequency[0] / SAMPLE_RATE);
>>> if(data->phase[0] > 1.0f) data->phase[0] -= 2.0f;
>>> data->phase[1] += (data->frequency[1] / SAMPLE_RATE);
>>> if(data->phase[1] > 1.0f) data->phase[1] -= 2.0f;
>>
>> You haven't shown us the decl
Hello,
Indeed, that was the cause - at 48 kHz sampling rate the output spectrum
is good. The sampling rate conversion is probably done inside ALSA -- when
accessing device 0 it worked as expected.
Thanks,
Eugen Epure
> Hi,
>
> Try using 48 kHz sampling rate instead of 44.1 kHz (or the other way
The factor in the sine argument must be 2*M_PI, otherwise it generates
half the frequency.
Because of the sine periodicity, all this work:
if(phase > 1.0) phase -= 1.0; // sine arg rolls over 2*PI->0
if(phase > 1.0) phase -= 2.0; // 2*PI->-2*PI
if(phase > 0.5) phase -= 1.0; // PI->-PI
First is pro
Might want to ask this one on the PA list (:
On Wed, Apr 27, 2011 at 2:47 PM, wrote:
> Hello,
>
> Today I tried compiling on Windows with MinGW & MSYS, and everything
> works, the spectrum is clean at SR=44.1 kHz and even the LUT version is
> acceptable.
>
> Probably on linux I was mixing differ
Hello,
Today I tried compiling on Windows with MinGW & MSYS, and everything
works, the spectrum is clean at SR=44.1 kHz and even the LUT version is
acceptable.
Probably on linux I was mixing different versions of header and linked
library. I also tried in linux copying directly input buffer to ou
that distortion.
Xue
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of Ross Bencina
Sent: 27 April 2011 06:39
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Sinewave generation - strange spectrum
eu
On Apr 27, 2011, at 1:38 AM, Ross Bencina wrote:
eu...@lavabit.com wrote:
*out++ = data->amplitude[0] * sinf( (2.0f * M_PI) * data->phase[0] );
*out++ = data->amplitude[1] * sinf( (2.0f * M_PI) * data->phase[1] );
/* Update phase, rollover at 1.0 */
data->phase[0] += (data->frequency[0] / SAM
eu...@lavabit.com wrote:
*out++ = data->amplitude[0] * sinf( (2.0f * M_PI) * data->phase[0] );
*out++ = data->amplitude[1] * sinf( (2.0f * M_PI) * data->phase[1] );
/* Update phase, rollover at 1.0 */
data->phase[0] += (data->frequency[0] / SAMPLE_RATE);
if(data->phase[0] > 1.0f) data->phase[0]
rule out everything except your algorithm ?
>>
>> Regards
>> Rob
>>
>> -Original Message-
>> From: Alan Wolfe
>> Sent: Tuesday, April 26, 2011 9:14 PM
>> To: A discussion list for music-related DSP
>> Subject: Re: [music-dsp] Sinewave gene
ginal Message-
> From: Alan Wolfe
> Sent: Tuesday, April 26, 2011 9:14 PM
> To: A discussion list for music-related DSP
> Subject: Re: [music-dsp] Sinewave generation - strange spectrum
>
> just stabbing in the dark in case nobody else gives a more useful
> response but...
Hi,
Try using 48 kHz sampling rate instead of 44.1 kHz (or the other way).
AFAIK the HDA Intel is an integrated soundcard, it might have a low
quality sample rate converter / interpolator.
Regards,
Gabor Pap
On Tue, Apr 26, 2011 at 11:55 PM, wrote:
> Thanks for the quick response.
>
> #1 Outp
Thanks for the quick response.
#1 Output format is paFloat32; I also tried initially with paInt16 and a
sine wavetable, and the result was almost identical.
#2, #3 I've just tried using doubles, and replacing data->amplitude with
0.1, and the result is identical, data->amplitude[0],[1] was consta
verything except your algorithm ?
Regards
Rob
-Original Message-
From: Alan Wolfe
Sent: Tuesday, April 26, 2011 9:14 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Sinewave generation - strange spectrum
just stabbing in the dark in case nobody else gives a mo
oh also, it *might* matter what device you are using with port audio.
for instance, when i use ASIO i seem to get a much louder, more raw
signal than if i use, say, the directsound interface.
i think the DS device must do some kind of dsp on it like maybe
there's a compressor or something.
To ru
just stabbing in the dark in case nobody else gives a more useful
response but...
#1 - what is the format of your output? If it's low in bitcount that
could make the signal more dirty i believe (less resolution to make a
more perfect sine wave)
#2 - have you tried calculating via doubles?
#3 -
Hello,
I want to generate two different frequency sinewaves on LineOut -
Left&Right. For audio IO I'm using Portaudio(Linux, PortAudio V19-devel
(built Apr 17 2011 22:00:29)), and the callback code is:
static int paCallback( const void* inBuff, void* outBuff,
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