igorzwx wrote:
> Have you noticed some audible noise in your video?
I am a radio developer. I do signal analysis with advanced methods and I note differences very much smaller than a human ear can distinguish. There is absolutely no difference at all between OSS and ALSA provided the user knows how to avoid resamplers and format converters as I described in my posting.
That is what I thought would be clear by this statement: "In both cases the data is the unmodified raw data supplied by the soundcard as is mandatory for a SDR"
The answer to your question is: There is absolutely no difference in the noise floor between OSS and ALSA. I would say the noise floor of my modified Delta 44 is not audible, but yet it is the limiting factor when the Delta 44 is used as part of a radio receiver. The noise floor is at about -145 dB in a 1 Hz bandwidth.
igorzwx also wrote:
> Have you tried to remove PulseAudio?
Sure, it affects the latency, but it does not affect the signal.
igorzwx also wrote:
> You may also try a "test for deafness":
> _http://ossnext.trueinstruments.com/forum/viewtopic.php?f=3&t=5831
On this page I read: "You may not hear any difference between HiRes and CD format with ALSA, simply because ALSA resamples everything with a low quality resampler. Since ALSA does not provide any sort of 'exclusive mode', it should not be used for hearing tests." THIS IS A FALSE STATEMENT!!!!! The way to run ALSA in exclusive mode is to probe the soundcard for modes supported by hardware. See the routine alsar_get_dev_native_capabilities(...) in lsetad.c of the linrad package.
On the page I also read: "What really matters is what the term 'exclusive mode' actually means. The true meaning of the term is that certain software causes of sound distortions are somehow removed:
1. any sort of software mixing and/or redirection to virtual mixer engines is disabled,
2. sample rate/format conversions are disabled,
3. any sort of dsp processing, such as equalizing, is disabled."
The purpose of the alsar_get_dev_native_capabilities(...) is to make sure these criteria are met. The routine uses things like:
resamp=1;
err=snd_pcm_hw_params_set_rate_resample(tx_ad_handle, hw_ad_params, resamp); err=snd_pcm_hw_params_set_rate_near(tx_ad_handle,hw_ad_params, &samprate, 0);
to make sure the original raw data from the soundcard will be sent to the callback
routine that takes care of the sound stream.
I might add that the Delta 44 when modified
http://sm5bsz.com/linuxdsp/hware/delta44.htm works fine under 32 bit Windows XP but that there is no proper routine for any 64 bit Windows.
It is possible to install the 32 bit XP driver under 32 bit Windows 10 by use of compatibility mode.
The Delta 44 works well in both 32 and 64 bit Linux under OSS as well as under ALSA.
The Lynx2 is totally useless because of the strong phase noise on the sampling clock
http://sm5bsz.com/linuxdsp/rxiq/rx2500lynx.htmas well as under Linux with both OSS and ALSA but that there is no proper drive routine
(This might evem be audible by ear.)
The Maya 44 has a problem. There is a time shift between the two stereo channels. This
time shift is different from time to time and that makes the card difficult to use.
It is difficult to find a properly working 4-channel soundcard....
I do however not think that "test for deafness" would make any of those very serious
errors audible.
Regards
Leif
Or Jakob Blumkin