Linux-audio-dev July 2010

linux-audio-dev@lists.linuxaudio.org

69 participants
72 discussions

Floating point processing and high dynamic range audio

by JohnLM

Hey, fellow devs! I'm a newcomer to this mailing list and I'm sure this has been discussed here at some point, but I'll ask anyway. I'm a bit better on programming side than on audio-tech side so... - Does all floating point formats mandate valid values in [-1.0, 1.0] range? - How does floating point (and perhaps fixed point/integer) values relate to dB notation? - If I muliply float by 2.0, do I get twice the sound pressure or twice the sound intensity, or something else? Well as for internal processing. If I mostly use integer formats for inputs, it doesn't appear I gain anything from "normalizing" them (scale to [-1.0, 1.0]) during internal processing. I just output integer part directly to integer formats (adjusted for different bit resolutions when necessary), and normalize only when saving to float format. What do you think? Proprietary formats such as AC3 and DTS market themselves as high dynamic range. Do they use values over 0dBFS or do they just have higher bit precision than "standard" formats? Thanks! -- JohnLM

14 years, 1 month

Re: [LAD] [LAA] announcing envy24control, mudita (*) edition.

by Jeff McClintock

> > Real detents require force feedback. A mouse doesn't > > provide that. This GTK thing is completely broken. Logitech iFeel mouses..would be cool if supported .. http://www.dansdata.com/ifeel.htm

14 years, 1 month

compiling hdsploader

by fons＠kokkinizita.net

Hello all, Anyone understands what is the problem here: fons@zita2:~/build/alsa-tools-1.0.23/hdsploader> make if gcc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"hdsploader\" -DVERSION=\"1.2\" -DSTDC_HEADERS=1 -DHAVE_LIBASOUND=1 -I. -I. -DDATAPATH=\"/usr/local/share/alsa/firmware/hdsploader\" -g -O2 -MT hdsploader.o -MD -MP -MF ".deps/hdsploader.Tpo" -c -o hdsploader.o hdsploader.c; \ then mv -f ".deps/hdsploader.Tpo" ".deps/hdsploader.Po"; else rm -f ".deps/hdsploader.Tpo"; exit 1; fi In file included from hdsploader.c:26:0: /usr/include/alsa/sound/hdsp.h:76:14: error: expected ‘:’, ‘,’, ‘;’, ‘}’ or ‘__attribute__’ before ‘*’ token hdsploader.c: In function ‘upload_firmware’: hdsploader.c:73:13: error: ‘hdsp_firmware_t’ has no member named ‘firmware_data’ make: *** [hdsploader.o] Error 1 TIA, -- FA There are three of them, and Alleline.

14 years, 1 month

Re: [LAD] twice as loud

by lieven moors

On 07/21/2010 07:24 PM, Fons Adriaensen-2 wrote: > On Thu, Jul 22, 2010 at 01:05:01AM +0200, Philipp Überbacher wrote: > > > I think the word loudness is a problem here. Afaik it usually refers to > > how it is perceived, and twice the amplitude doesn't mean twice the > > perceived loudness. It may mean twice the sound pressure level, energy, > > or intensity (if we ignore analogue anomalies, as you wrote in some > other > > answer). > > Subjective loudness is a very complex thing, depending on the > spectrum, duration, and other aspects of the sound, and also > on circumstances not related to the sound itself. > > For mid frequencies and a duraion of one second, the average > subjective impression of 'twice as loud' seems to correspond > to an SPL difference of around +10 dB. > > I often wondered what criterion we use to determine which > objective SPL difference sounds as 'twice as loud'. We don't > have any conscious numerical value (there may be unconscious > ones such as the amount of auditory nerve pulses, or the amount > of neural activity), so what it this impression based on ? Sorry to post this twice, but I think the mail got truncated. Hi Fons, I'm a fool to even try to answer this question. But I couldn't resist... Let's suppose we have two sounds A and B, and sound B has been measured as being twice as loud as A, by somebody. In order to be able to say that, that person needs some kind of reference measurement unit, the equivalent of a measurement stick. That unit has to satisfy two requirements. It has to be big enough, so that people can agree some difference is being measured, and it has to be small enough, so that a multiples of that unit fit into a realistic range. There is a requirement of maximum precision (the smallest value we can measure), and a requirement of minimum precision. The question is, what kind of measurement stick is being used by that person. First of all, we can assume that the length of that stick will be depend on the range of possible input values that we observe, and that we want to measure. If we want to measure the size of a road, we will probably use kilometers, instead of meters. In the same way, when our ears want to measure the amplitude of a sound, our ears will use smaller or bigger units, depending on the ranges observed. What are the ranges we observe? Let's assume that humans are perfect, and observe everything that we can observe with SPL meters. We could do a statistical investigation on a number of people, and make charts of everything they hear. In these charts we would see what frequencies they are exposed to, and what the minimum and maximum SPL's are for that frequencies. After more analyses, we would have one chart that could be representative for most people. >From that chart we could get an estimate of the size of the measurement unit. Frequencies with with bigger SPL variations would be measured with bigger units, and visa versa. And from this we could deduce what the minimum precision is for a certain frequency, when we say it is twice as loud. To satisfy the requirement of maximum precision, we should take into account the smallest observable differences for every frequency in the spectrum. now you can kill me :-) Greetings, Lieven

14 years, 1 month

Can someone add 2 features to Kluppe?

by Louigi Verona

Hello guys! I would like to ask any of the developers who might be interested to help a musician out! Original Kluppe developer seems very busy these days. And his programm lacks two things I really need for my musical work. I am ready to pay for the work. All the details below. *1. Feature 1 - random play.* When on Windows, I wrote a simple programm for myself, called Tape Loops. It could play sounds either once or in a loop. The special function which I added to it was "random play". What it did was allow you to define a period of time in seconds. And the programm would trigger the sound randomly somewhere within this period. So, if the period is 20 seconds, the programm would play the sound either in 5 seconds, or in 10 or in 7 or in 20. When the sounds stops playing, the timer is reset and the program once more chooses a random moment to trigger the sound within the given period. By changing the period the musician can make the sound get triggered more often or less often. The demo of how this program works can be viewed here: http://www.louigiverona.ru/?page=projects&s=software&t=tapeloops&a=tapeloop… It is a great feature - it helps a lot an ambient composer like myself and is greatly useful for installations. On Linux I have tried some programming, but even setting up JACK is very difficult for me, I am absolutely not a strong desktop programmer. So writing something like that from scratch on Linux is not realistic for me. I tried Kluppe, which is the closest thing, it is a great piece of software, but I studied the code for several days, tried some things, but apart from changing the colors, I do not seem to be able to do anything meaningful. So I would like to ask someone to do this job for me. To add a timer to a kluppe looper and to allow this "random play" mode, where the musician can put a looper into random play mode and define the period. *2. Feature 2 - basic midi control.* Looking through Kluppe code, I saw that a lot of midi is already done, but it is not "attached" to the controls. I might be wrong and there may be more work than it seems, but anyway. I would want to be able to assign midi control to triggering loops, volume and panning - at least that. Otherwise, Kluppe is very difficult to use in a live performance. However, instead of proposing to allow to create separate controls for each looper like they have in SooperLooper, I would advice (and actually, ask for this feature to be implemented in such a manner) to instead go for the Selected looper scheme. So that one would not need a dozen of knobs to control things. There should be an ability to have one "Selected" looper. Similar to what Dj Traktor Studio has. So you are binding midi not to a definite looper, but to the Selected looper, and thus you would require only two knobs (vol, pan) and three buttons (play/stop, Prev looper, Next looper). *3. Payment.* I understand that all of the above might be not as simple as it seems to me now. I would be willing to pay, as much as I can. I am able to pay through PayPal. I do not know how much money is a normal pay for such work, but I think something can be arranged. If I am not able to pay up instantly, I am willing to pay for several months in a row to cover the necessary expenses. I will also ask around on forums if someone will join me and also donate some money - while my random play function is probably too specific and is only something for me, midi control in Kluppe is something I believe many people would want. Thank you for your attention and I hope someone gets interested in the request! Louigi Verona. http://www.louigiverona.ru/

14 years, 1 month

A question about LV2 audio

by Louigi Verona

Hello everybody! When using an LV2 synth within a sequencer, I was told that the host has no way to access audio which is produced by the plugin so such a basic thing as rendering your whole project to wav appears to be impossible. I want to ask if this is true and whether this can somehow be changed. In my opinion, if the above is true, this is a very serious limitation of LV2. -- Louigi Verona http://www.louigiverona.ru/

14 years, 1 month

FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"

by Niels Mayer

I've been long annoyed by alsa-tools' envy24control (*) lack of peak-level indication in it's metering, and the non-implementation of its "Reset Peaks" button: https://bugzilla.redhat.com/show_bug.cgi?id=602903 I've fixed that annoyance. Now, a small white line appears above the highest level on each meter, and can be reset via "Reset Peaks" button. Original source: http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… Patch: http://nielsmayer.com/npm/602903.patch To run this version of envy24control grab the most recent stable release ( ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.23.tar.bz2 ) or git pull from trunk of the "alsa-tools" project ( http://alsa-project.org/main/index.php/GIT_Server ). After unpacking and assuming you've got the patch in ~/602903.patch do: >> cd alsa-tools-1.0.23 >> cat ~/602903.patch | patch -p1 it should give message "patching file envy24control/levelmeters.c" ... follow the directions to compile alsa-tools.... Can whoever's in charge of these things commit this change "upstream" and get it to propagate to the distros? Alternately, give me commit access and I'll do it myself, if people agree this patch is worth having. Next steps: (1) Think about building-in meter ballistics from Fons Adriaensen's jkmeter ( http://www.kokkinizita.net/linuxaudio/downloads/jkmeter-0.4.0.tar.bz2 ). Which helps implent Bob Katz' "K System" ( http://www.digido.com/level-practices-part-2-includes-the-k-system.html ). Although it probably most makes sense to implement such metering on the main "digital mixer" meter of envy24control, if it's not too inefficient, I'd want it available on all meters, despite the warning from Fons ( http://old.nabble.com/First-release-of-jkmeter-td18798950.html ). > This is the type of meter you want for live recording, > mixing and mastering. It probably makes no sense to > use it on all tracks of a DAW, where keeping digital > level within limits is the main purpose of metering. (2) Fix https://bugzilla.redhat.com/show_bug.cgi?id=602900 which causes 'envy24control' to crash when given "-D" argument. and behave incorrectly when given a correct non-numeric name for an audio device: e.g. >> envy24control --card M66 >> invalid card type (driver is USB-Audio) >> envy24control --card M66 --device M66 >> Segmentation fault (core dumped) (3) Apply more reasonable defaults and automatically size to correct dimensions. I use, for example: >> envy24control --card 2 --outputs 4 --input 4 --pcm_output 8 --view_spdif_playback --midichannel 1 --midienhanced --window_width 1275 --tall_eq_mixer_heights 1 & (4) Add midi output, in addition to midi input, which would monitor the hardware peak metering from the ice1712 ("amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45") and output a MIDI note-on on the corresponding MIDI channel whenever a specified threshold value was exceeded. This could be used to implement "automatic gain control" in association with the midi-controllable ADC gain provided by envy24control. Niels http://nielsmayer.com PS: (*) http://alsa.opensrc.org/index.php/Envy24control Cards supported by this fix: * M Audio Delta 1010 * M Audio Delta 1010LT * M Audio Delta DiO 2496 * M Audio Delta 66 * M Audio Delta 44 * M Audio Delta 410 * M Audio Audiophile 2496 * TerraTec EWS 88MT * TerraTec EWS 88D * TerraTec EWX 24/96 * TerraTec DMX 6Fire * TerraTec Phase 88 * Hoontech SoundTrack DSP 24 * Hoontech SoundTrack DSP 24 Value * Hoontech SoundTrack DSP 24 Media 7.1 * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton * Terrasoniq TS 88

14 years, 1 month

Solved: No Flash audio. With a question.

by Tim E. Real

Hello. Just a follow up to something I said a while back. I think Gene H. was saying he couldn't get audio and I had replied that I disabled Pulse and it appeared to be working. But even after disabling Pulse, sometimes I still was not getting audio in Flash on YouBoob etc. When I did have audio, I noticed that every video's audio seemed slower and lower in pitch. I was fooled into thinking that the owners of the videos had somehow remixed them at lower pitches or something. It turns out that Flash wants to play everything at a 48000Hz sample rate. Since my audio card is an Envy24 based card, it always boots up in 44100Hz sample rate mode. So now, before I play Flash videos, I must open the envy24control mixer and set the sample rate to 48000Hz. It works every time now ! Can someone explain the process of how Flash plays audio, and how to force it to use 44100Hz? I'm confused because at 44100Hz the audio is in sync with the video, yet it is lower in pitch and slower. Does this mean Flash is really playing the video slower? Thanks. Tim.

14 years, 1 month

twice as loud

by lieven moors

On 07/21/2010 07:24 PM, Fons Adriaensen-2 wrote: > On Thu, Jul 22, 2010 at 01:05:01AM +0200, Philipp Überbacher wrote: > > > I think the word loudness is a problem here. Afaik it usually refers to > > how it is perceived, and twice the amplitude doesn't mean twice the > > perceived loudness. It may mean twice the sound pressure level, energy, > > or intensity (if we ignore analogue anomalies, as you wrote in some > other > > answer). > > Subjective loudness is a very complex thing, depending on the > spectrum, duration, and other aspects of the sound, and also > on circumstances not related to the sound itself. > > For mid frequencies and a duraion of one second, the average > subjective impression of 'twice as loud' seems to correspond > to an SPL difference of around +10 dB. > > I often wondered what criterion we use to determine which > objective SPL difference sounds as 'twice as loud'. We don't > have any conscious numerical value (there may be unconscious > ones such as the amount of auditory nerve pulses, or the amount > of neural activity), so what it this impression based on ? > > The only thing I could imagine is some link with the subjective > impression of a variable number of identical sources. For example > two people talking could be considered to be 'twice as loud' as > one. But that is not the case, the results don't fit at all (it > would mean 3 dB instead of 10). > Hi Fons, I'm a fool to even try to answer this question. But I couldn't resist... Let's suppose we have two sounds A and B, and sound B has been measured as being twice as loud as A, by somebody. In order to be able to say that, that person needs some kind of reference measurement unit, the equivalent of a measurement stick. That unit has to satisfy two requirements. It has to be big enough, so that people can agree some difference is being measured, and it has to be small enough, so that a multiples of that unit fit into a realistic range. There is a requirement of maximum precision (the smallest value we can measure), and a requirement of minimum precision. The question is, what kind of measurement stick is being used by that person. First of all, we can assume that the length of that stick will be depend on the range of possible input values that we observe, and that we want to measure. If we want to measure the size of a road, we will probably use kilometers, instead of meters. In the same way, when our ears want to measure the amplitude of a sound, our ears will use smaller or bigger units, depending on the ranges observed. What are the ranges we observe? Let's assume that humans are perfect, and observe everything that we can observe with SPL meters. We could do a statistical investigation on a number of people, and make charts of everything they hear. In these charts we would see what frequencies they are exposed to, and what the minimum and maximum SPL's are for that frequencies. After more analyses, we would have one chart that could be representative for most people. >From that chart we could get an estimate of the size of the measurement unit. Frequencies with with bigger SPL variations would be measured with bigger units, and visa versa. And from this we could deduce what the minimum precision is for a certain frequency, when we say it is twice as loud. To satisfy the requirement of maximum precision, we should take into account the smallest observable differences for every frequency in the spectrum. now you can kill me :-) Greetings, Lieven

14 years, 1 month

STEREO RULES

by Ralf Mardorf

People today aren't able to do a good stereo or mono mix, e.g. because of the loudness war, but they are thinking of doing 3D mixes. I'm unable to follow this strange evolution. We all have 2 ears and 1 brain that has to do a lot of work, regarding to information from the sense organs. The brain needs to do math because usually the left and right ear are not equal, so even when wearing a head phone the brain needs the context of the situation and the perfect natural loudness etc. for this context. All thoughts about doing 3D are useless until we aren't able to connect electrodes directly to our brains and to have a rimming regarding to our brains. As long as Cochlear implant isn't good, any try to do 3D mixing is laughable, resp. a pain, just try to watch AND hear a modern film, it's a torture. There are some good mixes for stereo and mono, but at least I never heard a valid mix with more but one ore two channels.

14 years, 1 month

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Linux-audio-dev July 2010