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#rockbox log for 2013-05-16

00:00:54 Join jhMikeS [0] (~jethead71@rockbox/developer/jhMikeS)
00:01:37neozen(after correcting paths for aapt and dx in android/android.make, and shoving that apkbuilder script in the SDK/tools directory)
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01:39:12webguest17how do i compile a rock file?
01:39:27webguest17i can compile rockbox
01:39:46webguest17but i dont know how to compile a single file
01:40:16webguest17i am a a programer but new in ubuntu
01:41:43jhMikeSyou can't. it's 'make bin', 'make rocks', 'make codecs'
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01:43:53webguest17you mean i can only compile all the rock files?
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02:06:29jhMikeSwebguest17: aye
02:09:11webguest17another question
02:09:34webguest17can i edit the main menue in rockbox
02:09:50webguest17i want to remove radio from the menue
02:11:32jhMikeSyou can modify the appropriate c file but menus are not configurable and it has been agreed they shouldn't be
02:11:56webguest17its just for my kid
02:12:44saratogayou can actually remove items from the very top level menu
02:12:54saratogaits in the manual
02:12:56webguest17which will bel appropriate c file ? under "firmware"
02:13:19jhMikeSsaratoga: wow, news to me even, lol
02:18:41webguest17ok! ive got it root_menu
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02:35:08[Saint]So, its no secret I'm not much of a sound guy, so. Does a precut of -6dB == the equivalent of 0dB at full volume if +6dB is the max?
02:35:31[Saint]Or, does it just not work like that and it'd clip past 0dB regardless of the precut anyway?
02:39:17LloreanThat's the way I thought it works, but I'm no expert either.
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02:53:49[Saint]Llorean: errrr, just to be specific, which? The former, or the latter?
02:54:03[Saint]That it works as I expected, or that it'll just clip anyway?
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03:00:23jhMikeS+6dB is over-unity
03:00:57jhMikeSso, precut -6 will net +0
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03:20:47saratogathat'll lower the amplitude by a factor of 2 digitally, and then raise it by a factor of 2 after the DAC (in analog)
03:30:05 Quit krabador (Ping timeout: 260 seconds)
03:35:57soapthere are two things I don't grok in that, saratoga.
03:36:18soap1 - won't it lower it by (roughly) a factor of 4?
03:36:35soap2 - and what's this about it raising it again in the analog domain?
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03:44:07saratoga_soap: it'll raise the amplitude 2x and the power 4x
03:44:13saratoga_since power is amplitude squared
03:45:57saratoga_its why theres two formulas for dB, 20*log10(voltage) and 10*log10(power)
03:47:07saratoga_analog gain just means it turns the power up on the amplifier, whereas digital means the the bits are shifted over 1 place so that the DAC produces a smaller voltage
03:48:19saratoga_lowering the digital gain makes sense if you're going to then apply some digital effect that will increase the level (say positive EQ) and you want to make sure you don't clip
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03:50:24saratoga_i guess there isn't ever really a reason to need postive analog gain in rockbox, but its handy if you don't feel like enabling the EQ precut or replaygain just to raise the level of some quiet tracks
03:55:06soapok, saratoga, question 1 answered.
03:55:18soapquestion 2 of mine apparently wasn't worded well.
03:55:24soapLet me rephrase:
03:56:33soap"What I get get from your original response is that using the -6db precut trick will in and of itself lower it 6dB in the digital domain AND raise it back up 6dB in the analog domain". Is that what you were trying to say?
03:57:19saratoga_no the precut lowers it 6dB, then the volume setting raises it back up 6dB
03:57:31saratoga_so there is no net change in volume
03:59:36soapso an iPod at volume 6 outputs the same voltage as an ipod at volume 6 with a precut of -6?
03:59:47soap(which was [Saint]'s query)
04:01:13saratoga_oh wait no
04:01:49saratoga_that wasn't his question
04:02:06saratoga_he asked if -6dB then +6dB = 0 dB
04:02:44soapand you appear to be saying that the precut does not accomplish that
04:02:50saratoga_an ipod at +0dB, +6dB will have 2x the amplitude (or clip trying)
04:04:28saratoga_"that'll lower the amplitude by a factor of 2 digitally, and then raise it by a factor of 2 after the DAC (in analog)"
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10:49:30ukleinekmoin, I'm looking into firmware/target/arm/imx233/audioout-imx233.c because I have to write a Linux driver for it.
10:50:02ukleinekthe timeings copied from the HRM are not optimal.
10:50:07***Saving seen data "./dancer.seen"
10:51:34ukleineke.g. for HW_FREQ_11 { 0x1, 0x3, 0x11, 0x38 } is a better value
10:54:13ukleinekthe value in the driver/reference yields 11025.057601619304 while my values yield 11024.978466838931.
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10:55:10wodzukleinek: thats the remark for pamaury
10:55:24ukleinekwodz: thanks
10:55:29wodzmortalis: You should add MA9 to mainpage
10:56:13pamauryukleinek: I took the values from the datasheet iirc (I wrote a general code too but left it commeted), thanks for noticing, i'll check later
10:56:49ukleinekpamaury: ah, I see
10:57:35ukleinekpamaury: for 192000 the value in the datasheet is plain stupid. You could get an exact match but the suggestion fails to do that.
10:58:15mortalisyes, i remember, i will do it later today
10:58:19pamauryto be honest I didn't care too much because rockbox uses 44100Hz for virtually anything ^^
10:58:39*ukleinek checks HW_FREQ_44
10:59:39pamauryinterestingly I recall writing some code to find the value best values (with brute force) and concluded that the values of the datasheet were indeed the best, maybe my memory is faulty
10:59:51ukleinekpamaury: datasheet -> 44100.23040647722, with s/37/38/ 44099.913867355725
11:00:27wodzpamaury: I was thinking about the way to recover your rk27generic DAP. The easiest way seems to be this
11:00:47pamaurywodz: thanks, i'll have a look
11:01:01ukleinek..ooOO(much homework for pamaury :-)
11:02:08wodzpamaury: The other possibility I can think of is to overwrite nand loader with the image grabbed from my DAP.
11:02:08pamauryukleinek: are you sure there is no trick like dividing by SRC_FRAC+1 instead of SRC_FRAC ?
11:02:28ukleinekpamaury: also I wonder why dacssr doesn't just contain a u32.
11:02:43wodzpamaury: The downside is that I am not quite sure in which way ftl is screwed so I don't know if this will work at all
11:03:25pamaurybecause it's more complicated than just a divider, you can adjust oversampling in the internal paths for example and there are fractional dividers
11:03:47wodzpamaury: the files for RM970 should work (at least I think so)
11:04:36ukleinekoutput_sample_rate = (6x10^6 * BASEMULT) / (SRC_INT.SRC_FRAC * 8 * SRC_HOLD + 1))
11:05:37ukleinek= 75e4 * basemult * (1 << 13) / ((SRC_INT << 13) + SRC_FRAC * (SRC_HOLD + 1))
11:06:35ukleinekwell, I screwed the parens
11:07:20DBUGEnqueued KICK ukleinek
11:08:30pamauryok, i'll have a look at this later, i'm quite busy at the momentn and i'll tell you if I confirm or if I found an error :)
11:08:45ukleinekpamaury: very appreciated
11:09:16pamauryif you understand some of the audioout parameters, I am very interested because I can't get rid of background noise/hiss on the fuze+
11:13:15CtcpIgnored 1 channel CTCP requests in 0 seconds at the last flood
11:13:15*ukleinek works on
11:14:20pamauryukleinek: I have a chumby infocast 3.5" ! I've begun a rockbox port, not finished though
11:16:26ukleinekpamaury: my formula is:
11:16:30ukleinek div13 = SH_DIV(750000 * basemult, outrate * (hold + 1), 13);
11:16:30ukleinek divfrac = div13 & ((1 << 13) - 1);
11:16:30ukleinek divint = div13 >> 13;
11:17:07pamauryukleinek: I just dug out of my hard drive this code I told you about to find the best dividers: pamaury/5590501">
11:17:23pamauryYou can have a look at it while i'm busy, maybe you'll find my error or yours or none ^^
11:18:09ukleinekpamaury: /me takes a look
11:18:27pamauryI wrote it two years ago or so, I don't remember any of it
11:23:56ukleinekyou're using unshifted integer math, so 44100.23040647722 and 44099.913867355725 is both 44100 for you
11:27:05pamaurymake sense
11:41:29pamaurywhat is UWE ?
11:41:46ukleinekpamaury: my first name
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11:42:11pamauryok ^^ thanks
11:42:19pamauryi'll need to update the table in our code then
11:43:27ukleinekin return I'll copy from rockbox instead of a vendor kernel from the middle age
11:44:26pamauryI'm interested in some feedback about anti-pop code, the best way I found was this delay before unmuting. Also my code doesn't handle the speaker yet
11:45:50pamaurybut i have a device with a speaker
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11:47:28ukleinekpamaury: /* minimum is -100dB and max is 0dB */
11:47:28ukleinek vol_l = MAX(-200, MIN(vol_l, 0));
11:47:43ukleinekthere is a mismatch between comment and code, isn't there?
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11:50:03pamauryukleinek: no no ^^ the hardware has half-db precision
11:52:05pamaurybasically internally rockbox uses thenth-of-db and uses a function (tenthdb2master) to go from tenth-of-db to internal representation, my internal representation is in half-db so *2 basically
11:52:20pamaurymight be a bit confusing
11:52:36*ukleinek nods
11:53:10pamauryalso I suggest you have a look at the code on my personal branch here: pamaury/rockbox-1/tree/imx233-renaming">
11:53:34pamauryit is more up-to-date and will eventually find its way to the trunk, it is a major rework of the port in many ways
11:53:43pamauryaudioout is not that different though
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11:54:35ukleinekpamaury: s/biais/bias/ still exists in the rewrite
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11:55:18pamaurybias you mean ?
11:56:15pamauryah ok, understood ^^ thanks
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11:56:44Mode"#rockbox +o Torne" by ChanServ (ChanServ@services.)
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11:57:41Mode"#rockbox +b krabador!*@*$##fix_your_connection" by Torne (~torne@rockbox/developer/Torne)
11:58:06Tornefreenode is magical
11:59:00ukleinekTorne: because they kicked him before you issued the command?
11:59:02pamauryukleinek: also, very very very important, maybe you can help me on this point
11:59:15ukleinekpamaury: what is IMX233_SUBTARGET?
11:59:41pamaurymagic value to handle stmp3600, stmp3700 and stmp3780/imx233 at the same time, for your purpose, IMX233_SUBTARGET would be 3780
12:00:37Torneno, because $##fix_your_connection is a neat trick that most ircds don't have :)
12:00:38pamauryif you look at pcm-imx233.c, you'll notice a comment about dma channel, that's something I never sorted out: it seems that if the audioout is running and you reset the dma channel, it just breaks everything and you are good for a complete audioout reset. Maybe i'm missing something
12:00:46Mode"#rockbox -b krabador!*@*$##fix_your_connection" by Torne (~torne@rockbox/developer/Torne)
12:01:19ukleinekpamaury: will keep that in mind.
12:01:28pamauryin theory it would only clear RUN and thus clkgate audioout as datasheet says but in practice that's not what happen on my device
12:01:49pamaury(and this is not the only explain of this problem)
12:09:12pamauryukleinek: last piece of advise, maybe your are aware of it. I haven't implemented speaker yet (I'll do it) but I seem to recall some linux/freescale/sigmatel code which states that to "power up" the speaker you have to send a few samples to it at the beginning otherwise it would stay silent.
12:10:09*ukleinek nods
12:10:26ukleinek... and leaves for lunch
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12:56:21Mode"#rockbox -o Torne" by Torne (~torne@rockbox/developer/Torne)
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14:24:39pamauryhum, this is weird: freescale bootloader sets HW_POWER_DEBUG_*LOCK fields, which are marked as reserved, which basically screw up the VBUSVALID irq and I have absolutely no idea why those are set
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14:28:20ukleinekpamaury: /* change biais to -50% */
14:28:20ukleinek BF_WR(AUDIOOUT_TEST, HP_I1_ADJ, 3);
14:28:30ukleinekaccording to the datasheet 3 = +50%
14:29:23pamauryDoesn't the comment apply to the next line, I don't remember why I put this, probably because the fuze+ OF (based on Freescale SDK) does this and it improves audio output
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14:30:03pamauryhonestly I don't understand exactly what those bits do, I'm no electrical engineer
14:30:54pamauryhum you're right there is a mistake here, the comment must be wrong
14:31:23*ukleinek agrees, I'm also no electrical engineer.
14:36:05pamauryI found this in the stmp3600 sdk, it may or may not apply, still better than nothing: pamaury/5591411">
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15:12:31[Saint]saratoga: soap: I read the backlog, I promise...but, I *know* my question was answered, but now I'm sitting here completely unsure as to what the answer actually was.
15:13:45[Saint]SO, if it can just be a yes or no answer, does a volume of +6dB with a precut of -6dB == 0dB? If its not that simple that's ok.
15:14:29Tornepretty sure it's that simple
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15:15:20*Torne reads the thing above.
15:15:32Torneyeah, it's definitely yes.
15:16:03[Saint]Right. That's what I thought. But after reading and re-reading the above I was still second guessing it.
15:17:33[Saint]I was just after a solution to stop Ms [Saint] workmates from setting ridiculous volume levels that will almost certainly be clipping because they don't understand the principle of line level.
15:17:54Torneyeah, it works fort hat, or for any kind of volume limiting in general.
15:18:59[Saint]I'll add a 6dB precut to fixed.cfg, thanks.
15:20:16[Saint]I suppose I could just set the max volume level as 0dB in my build, but I'm /almost/ back at a vanilla tree again and I don't want to go reinventing wheels.
15:21:29[Saint]The tiny CPU hit from having the EQ on with all bands at zero is absolutely irrelevant on the Classic.
15:22:33[Saint]well, in all fairness, having all ten bands with non-zero values on the Classic is an almost irrelevant CPU hit too.
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17:17:35ukleinekpamaury: the chumby sets HP_I1_ADJ to 1
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17:32:32Maxdamantusis there a a function somewhere to shuffle the current file/database list, or if not, would it be difficult to program?
17:33:08Maxdamantusto avoid playing music by artists with names starting with "A"
17:34:00melmothXMaxdamantus: create a playlist and shuffle that, or I am understanding it wrong?
17:34:22Maxdamantusin the actual file/database viewer.
17:34:42melmothXwell, when you play something, you are inside a playlist, so reshuffling should work
17:34:45Maxdamantusso if you go into Database→Artists, it's all alphabetical.
17:35:06Maxdamantusso I'm more likely to select someone whose name starts with A or B.
17:35:25melmothXwhy the listing should be random?
17:35:53Maxdamantusso I don't tend to select music by artists whose names start with A or B.
17:36:19Maxdamantusmaybe just being able to jump to a random position would be enough.
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17:37:16MaxdamantusI usually select all music by a particular artist, but don't have an artist in mind when going to select.
17:37:33Maxdamantusso I'll scroll down the list until I see something I'm in the mood for.
17:38:16melmothXlol, start from the bottom, or close the eyes and keep the down button pressed for a bit :-)
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17:39:39Maxdamantusyeah, neither way results in a uniform random distribution .. I'll just add a function to jump to a random position.
17:39:53Maxdamantusshould be simple enough.
17:40:21Maxdamantushmm .. unless I can make it loop while holding the down button.
17:41:26Maxdamantustakes about 6s to scroll to the bottom with my current settings.
17:42:16Maxdamantusand the probability of hitting something at the end (other than the very last item) is far lower than hitting something at the start, assuming my selection from 6s is uniform.
17:42:25Maxdamantus(because of acceleration)
17:43:12Maxdamantusactually, maybe it'd be nice to change the default behaviour of the current <Random> item.
17:43:40Maxdamantusmeh, probably not.
17:44:28Maxdamantuswas going to say: instead of going back to <Random> when leaving the randomly selected artist, go back to the artist (item in the Artists list)
17:48:30pamauryukleinek: so maybe my comment is right and I made a mistake when writing the value
17:50:03pamaurycopper: ping
17:50:44copperpamaury: hai
17:51:14pamaurycan you do a build with a modification and check if the background noise is gone on the fuze+ ?
17:51:28coppercompile you mean?
17:51:45copperturns out I didn't set a build env for rockbox after all
17:51:58copperI just used your compiled rockbox.sansa
17:52:22pamauryhum ok, then I'll try to send you one later to compare
17:52:47pamauryukleinek: apparently this is a mistake since in trunk I set it to 1 and not 3
17:53:18copperpamaury: also,
17:53:37coppersome EEs concluded SOMETHING that I'm not sure I understand
17:55:33copperI think the conclusion is, the noise exists, but it's below 16bit noise floor
17:55:46copperbut because of IEM high sensitivity, it's audible
17:56:06copperI still have a hard time wrapping my head around the idea that I'm hearing anything below 16 bit noise floor
17:56:42copperdoesn't really compute
17:57:21pamauryi can hear something too
17:57:34ukleinekpamaury: re your rate setting values, I think the relevant difference between your and my version is just that I round in the computation of src_frac
17:57:40pamauryIf I correctly this maybe be related to the voltages ? (I cannot make sense of it either)
17:58:59pamauryukleinek: make sense, since my code is not aware of the fractional part it rounds good but not necessarily best
18:00:28ukleinekpamaury: so maybe just add another "+ f * (src_hold + 1) / 2" on the left side of /
18:01:30pamaurycopper: If I understand correctly some of the setting of the DAC and what they do, we could try to lower the voltage, that would reduce overwall volume but the DAC will be more robust to small vddd perturbation, need to check that
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18:01:59copperis that a new MST?
18:02:09pamauryhehe, digital voltage
18:03:39pamauryanalog voltage (vdda) is derived from digital voltage (vddd) and apparently perturbation in vddd (like noise from the switching converter which powers the chip) can interfere, not sure this is the problem here but that's worth giving a try
18:05:03pamaury(this is not 100% technically correct but that's not important)
18:08:55saratogais gerrit searchable for keywords?
18:09:09saratogai tried "ipod" in the search box and it just gives me an error
18:10:30pamaurysaratoga: yes, but I don't remember how you do this, the search box at the top doesn't work
18:13:46saratogaso its "message:[text]" to search
18:14:02saratogaweirdly "subject:[text]" is an error
18:14:13Tornethat's because "subject" is just a convention, not a real field
18:14:19Tornegit commits don't really habe subjects
18:14:23Torneit's just a convention that the first line is one.
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18:33:57copperpamaury: I'll be back in ~20 minutes
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18:52:24fs-bluebotBuild Server message: New build round started. Revision a7dee7f, 214 builds, 22 clients.
18:59:57fs-bluebotBuild Server message: Build round completed after 454 seconds.
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19:48:23madcat1990yay new build! *checks log*
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20:11:39amayerwill the new hermite polynomial resampler effect battery life at all?
20:13:24jhMikeSDepends on your file encoding, and probably not much at that.
20:14:56jhMikeSIf your files are 44.1kHz, it's not used unless you're changing pitch.
20:16:15amayerwhat about increasing playback speed
20:16:32amayeri change that a lot under the pitch menu when listening to podcasts
20:16:34jhMikeSfor simple timestretch no
20:17:01amayerjhMikeS, thanks
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20:28:12madcat1990I'm hearing some weirdness in the background with that new commit on my ipod mini 2g
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20:29:16madcat1990wait nvm, it's nothing
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21:50:56ukleinekpamaury: when does the "pop" occur? When unmuting? (log)
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22:52:27pamauryukleinek: yes
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23:06:42ukleinekpamaury: do you have the hardware reference manual?
23:07:06*ukleinek has Rev 1 from 11/2009
23:07:11 Quit amayer (Ping timeout: 245 seconds)
23:08:10ukleinekpamaury: see page 1220
23:08:15pamauryactually I have two versions of it: rev 1 11/2009 and rev 4 03/2009
23:08:33ukleinekwow, why I don't have rev4?
23:08:45 Quit Galois (Ping timeout: 246 seconds)
23:09:02pamaurybecause it's difficult to find and it's actually pretty confusing since it's doesn't look like a rev 4 to me
23:09:49pamaurywhat about page 1220 ?
23:10:15pamauryI can send you the rev 4 if you want
23:10:26ukleinekpage 1220 describes pops
23:11:59pamauryyes and there is page 1230 but if you can make any of those techniques work, please tell me
23:12:42pamauryI personally find it dubious to have one anti-pop procedure described in a reserved bit and the one in audioout talking about rtc persistent bits which really seem unrelated
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23:14:30pamauryand there is an inconsistency in the datasheet: it says you must ramp the volume but volume starts in a state which means "zero" and explicitely states and go in/out of it will cause a pop
23:15:05pamaurys/and/that if you
23:15:32pamauryactually even with lowest volume, unmuting will cause a pop on my fuze+ iirc
23:17:13*ukleinek would be happy to hear a pop, so I cannot say anything yet
23:18:46pamauryyour driver doesn't work ?
23:19:50pamauryrandom things to check: ungate analog clock (see clkctrl), unclkgate audioout obviously, power up dac and random bits in audioout_pwrdn
23:20:17pamauryungate headphone/speaker gate with a gpio ?
23:20:24pamaurymany device have this
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23:22:31pamaurynote that if you use a speaker you have to power up the speaker too in pwrdn, and unmute speaker in speakctrl
23:23:06pamauryand if you use a speaker there is this "magic" sequence you might have to send to the dac otherwise the speaker will not start for some reason
23:24:14ukleinekpamaury: I compared the audioout registers to the state of the of with no differences that should matter
23:24:47pamaurydid you check if the OF uses a speaker/headphone gate controlled by gpio ?
23:24:49ukleinekaudio clk in clkctrl_xtal is ungated
23:25:21ukleinekI'm not that far yet. DMA isn't running I'd say
23:26:35pamaurydma is not required to produce sound actually since you can push samples directly, but you need to set RUN obviously
23:27:01ukleinekI set RUN, but when I check the registers then, it's off
23:27:13ukleinekHW_AUDIOOUT_DACDEBUG = 3
23:27:37pamaurydid you push samples to HW_AUDIOOUT_DATA ?
23:28:07ukleinekdo you think that would work with run unset?
23:28:49pamauryprobably not but many the dac won't start if you set run without sending anything to the fifo so I would set RUN and then write a few samples to see
23:29:56ukleinekdo you have a suggestion what to write to DATA?
23:30:27pamauryzero or all ones or try a sin
23:31:32pamaurytry to set the volumes to something reasonable too
23:32:14ukleinekI hardcoded them to the values of the of
23:33:47ukleinekRUN keeps on now
23:35:34*ukleinek also has FIFO_UNDERFLOW_IRQ
23:36:26pamauryunderflow is not a problem, if the dac runs out of sample if will just wait you send more
23:38:51pamauryukleinek: what is the value of vdda and vag ? there are constraints to make the dac work
23:39:25ukleinekhow do I find out?
23:39:42ukleinekah, I remember setting that somewhere
23:40:19pamauryyou can print the fields related to vddda in power registers
23:41:27ukleinekpower registers?
23:42:03ukleinekI set: AUDIOOUT + 0x00000080: 0x03381030
23:42:57ukleinekHW_POWER_VDDACTRL = 0008270a
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23:48:08pamauryukleinek: why do you set LOW_PWR in REFCTRL ?
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23:49:37oldGood night all
23:49:51ukleinekpamaury: because the of does it, too
23:50:11oldI have rockbox problems
23:50:18oldthis with my eq
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23:50:41pamaurysame for XTAL_BGR_BIAS, not sure what is the effect, but if the OF works with...are those bit set even when playing sound ?
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23:50:53ukleinekpamaury: yes
23:51:15ukleinekI dumped the registers while the device was playing
23:51:32oldi have live music and try right setting sound
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23:51:54ukleinekdifferences are: CTRL 001f0043 <> 001f0040
23:52:04ukleinekDACSRR because I play a different sound
23:52:12oldbut i have problems with voice khz
23:52:14ukleinekDACVOLUME because the pending bits are set
23:52:24pamauryhum, so RUN is not set ?!
23:52:26ukleinekDACDEBUG 00000000 <> 00000003
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23:53:19ukleinekright, at least after I set RUN for the first time. I can set it in memedit though
23:53:47ukleinekPWRDN 0 <> 00000100
23:53:48oldmain question say please in how queue need setting any band ?
23:54:05oldbecause in eq have three parametrs
23:54:15ukleinekTEST 00000000 <> 00800000
23:54:20ukleinekDATA is different
23:54:29ukleinekSPEAKERCTRL 0 <> 01800000
23:55:16pamauryukleinek: you have MUTE set in SPEAKERCTRL !
23:56:24ukleinekright, does it matter? ok I won't hear anything, but that doesn't explain everything
23:56:29*ukleinek still drops MUTE
23:57:10oldWho can tell how right do setting equalizer please answer. I hope in your help.
23:57:31ukleinekpamaury: doesn't make a difference
23:57:39pamaurywhat are you trying to obtain exactly ?
23:58:05pamauryold: if someone knows the answer it will answer, otherwise try asking on the forum
23:58:07gevaertsold: not that many people know the eq well. If nobody answers here, you might have better luck on the forums
23:58:17ukleinekfinding out why run is cleared
23:58:47pamaurytry to setup a dummy dma transfer which loops and send a few samples

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