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 rule this out, what you can do is write your output to a file instead of (or in addition to) spitting it out to the speaker, then run your analysis on the sound file. Libsndfile is a nice, simple library for reading/writing sound files: http://www.mega-nerd.com/libsndfile/ On Tue, Apr 26, 2011 at 1:14 PM, Alan Wolfe <alan.wo...@gmail.com> wrote: > 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 - what is data->amplitude... does that ever change or is it just a > one time set volume adjustment for the left and right channels? > > On Tue, Apr 26, 2011 at 12:57 PM, <eu...@lavabit.com> wrote: >> 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, >> unsigned long frpBuff, >> const >> PaStreamCallbackTimeInfo* tInf, >> PaStreamCallbackFlags flags, >> void* userData ) >> { >> int16_t i; >> audioData* data = (audioData*) userData; >> float* out = (float*) outBuff; >> >> /* Prevent warnings */ >> (void) tInf; >> (void) flags; >> >> for( i=0; i<frpBuff; i++ ) >> { >> *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] -= 2.0f; >> data->phase[1] += (data->frequency[1] / SAMPLE_RATE); >> if(data->phase[1] > 1.0f) data->phase[1] -= 2.0f; >> } >> >> return paContinue; >> } >> >> When I checked the output spectrum for a 10kHz frequency using baudline >> (running on another PC), I got this http://images.cjb.net/80af2.png . The >> spectrum is clean only for output frequencies below 2-3 kHz. >> >> The tone generator inside baudline gives a clean spectrum at 10 kHz: >> http://images.cjb.net/b943b.png . >> >> What method would you recommend for generating a clean sinewave at 5-12 kHz? >> I think there is a bug somewhere, because the sine is computed in float >> for each sample and should be precise enough... >> >> Thanks >> >> >> >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, dsp >> links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp >> > -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
