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#rockbox log for 2021-07-31

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07:02:59 Quit munkis (Ping timeout: 252 seconds)
07:11:02yangThere seem to be different "qualities" of FLAC files? However I read once, that every FLAC file can be losslessly transcoded into .wav to create a CD disc. So, why are then different qualities of FLAC files ?
07:17:05gevaertsYou can losslessly transcode between wav and flac at will, but they still have a sample rate, bit depth and number of channels. If you get a 11025Hz 8 bit mono flac file, it transcodes to a 11025Hz 8 bit mono wav file, and that's not what we know as "CD quality"
07:17:37gevaertsFlac does not introduce quality loss, but you can lose quality before encoding...
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08:40:18yanggevaerts: thanks for your explanation
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09:48:05_bilgusI'm confused I was under the impression in most cases using a const int for a buffer is perfectly fine
09:49:06_bilgusBUT const int bufsz = TAG_MAXLEN+32;
09:49:06_bilgus char buf[bufsz] is ~900 bytes more code than char buf[TAG_MAXLEN+32];
10:21:31dconradoh my god I don't believe it
10:22:05dconradthe eros q's play/pause clicking can be fixed by... adding a -2 dc offset?
10:22:16dconradno way that's true...
10:23:25dconradand also the whole low volume clicking thing was.... because it wants signed data centered around zero, and we were sending data centered around pcm_factor_unity/2?
10:24:26dconrad... is that also true for the hosted port I wonder
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10:28:19dconradoh my god yes it is
10:28:29dconradspeachy, any thoughts on the above ^^ ?
10:31:40rb-bluebotBuild Server message: New build round started. Revision 7413442411, 303 builds, 10 clients.
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10:42:13rb-bluebotBuild Server message: Revision 7413442411 result: All green
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11:10:26dconradso now that it looks like both the standard and alternate volume scaling methods are viable for the eros q native, I wonder if there is a sound quality difference between them
11:11:30dconrador if the alternate is better as it future-proofs it if we ever get more bit depth to the dac
11:37:14speachy_bilgus: what do you think about adding a hook to gerrit that rejects patches with bad whitespace?
11:37:44speachy(or at least refuses to submit them)
11:38:51_bilgussounds good but I don't know how often it really happens
11:42:59speachydconrad: that's an interesting discovery
11:43:10 Quit massiveH (Ping timeout: 265 seconds)
11:43:22speachyI suppose this is really a PCM data format thing..
11:45:19speachyso you're saying our pcm internals are U16, centered at MAX_U16/2, but the ALSA is expecting S16 centered at 0
11:45:33speachyand the erosQ (and perhaps other codecs) are the same?
11:46:02speachyin our alsa code we can just as easily request U16/U32 formatting
11:46:12speachyor is this purely an issue with the volume control?
11:47:51dconradwell, it would appear the input to the sample scaling function is signed, centered about 0, and even though the function output is signed, we recenter it about PCM_FACTOR_UNITY/2
11:48:08speachyok. that's clearly jacked.
11:48:25speachywonder what other targets use softvol control.
11:48:27dconradwhat alsa wants I don't know, but the dac appears to want signed data ("two's complement", per the datasheet)
11:48:49dconradso I'm unsure if that's by design or an error
11:49:30dconradand, if I understand correctly, that means we really only get 8 bits of range before it clips?
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12:17:56speachyhmm. might improve performance on the hosted targets if we use S24 vs S32 in alsa.
12:19:14dconradwon't that just add an extra processing step to truncate it? or rework the scale factor logic, which is magic to me haha
12:29:49dconradI just can't believe the play/pause clicking was solved so easily... I was really worried it was an inherent hardware flaw that would be impossible to fix
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13:29:11dconradspeachy: by the way, do you have any interest (or maybe you already have...) in trying out the native eros q port? I think mine is the only one it's been tested on
13:38:48speachyyes, but much like the m3k, I've had to put my spare time into other pursuits lately
13:39:21speachy(I have two Qs here, one with the v1.2 and the other with the v1.3 OF)
13:41:00dconradif you wanted to put the native port on one of those, it's a pretty quick process, and it shouldn't care which version the OF is
13:42:39dconradonly downside is no dual boot yet
13:45:30dconrad(and you'll probably want to apply either g#3633 or g#3623)
13:45:33rb-bluebotGerrit review #3633 at : ErosQ Native: Add DC Offset to PCM data by Dana Conrad
13:45:34rb-bluebotGerrit review #3623 at : Eros Q Native: Alternate 16-bit volume scaling by Dana Conrad
13:53:41speachyfirst things first... back outside to deal with more household duties
13:56:21dconradhaha yeah I get that
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15:47:31braewoodsspeachy: i do know one thing impacts performance. dmix. i assume it's already disabled but it should be if it isn't.
15:47:57braewoodsit adds extra overhead which is pointless for a single audio source setup
15:54:37braewoodsoh right. i forgot. rockbox does have one kind of "sound effect"
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16:41:35yangWhich iRiver player is this ?
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18:00:30saratogadconrad: PCM_FACTOR_UNITY/2 is equal to 1/2
18:00:42saratogaadding one half and then truncating is a common way to implement rounding
18:03:42saratogayeah, looking at your patch you removed the rounding to the nearest integer
18:04:00saratogaif truncation sounds better than rounding you probably have the fixed point format set wrong
18:09:10saratogashould you be editing pcm_sw_volume.c at all? I think you can do whatever format conversion is needed in the individual device driver (or whatever the equivalent is for hosted devices), putting in device-specific workarounds into the general software volume control seems very messy and probably not needed
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18:50:13speachywon't adding PCM_FACTOR_UNITY/2 screw up rounding since the sample is signed?
18:50:25speachy(I mean, if the value is < 0, you need to subtract UNITY/2, not add it..
18:51:30braewoodsyang: not any supported one.
18:56:25_bilgusspeachy thats a good point
18:57:12_bilgusif you did u + val it would word
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19:07:30speachydconrad: please try g#3634
19:07:32rb-bluebotGerrit review #3634 at : pcm: Fix incorrect softvol rounding by Solomon Peachy
19:09:11speachythis makes me wonder if we have similar issues lurking elsewhere in our codebase
19:17:46 Join dconrad [0] (~dconrad@
19:54:18dconradspeachy, unfortunately g#3634 seems to crackle quite a bit
19:54:20rb-bluebotGerrit review #3634 at : pcm: Fix incorrect softvol rounding by Solomon Peachy
19:56:37dconradhowever, if I swap the signs on the conditional so it's "? -(PCM_F_T)PCM_FACTOR_UNITY/2 : (PCM_F_T)PCM_FACTOR_UNITY" it seems to not crackle/click at all, including starting/stopping/play/pause etc.
19:57:51dconradunfortunately my DC bias trick doesn't work to get mute to be silent with this patch, it clicks going in/out of mute
19:58:21dconradmy bad on my understanding of what PCM_FACTOR_UNITY is, btw
20:00:37speachyit does seem that the rounding that pcm_scale_sample() does is at the root of this issue.
20:00:57braewoodsspeachy: is this like the different FP rounding methods?
20:01:22braewoodsround even/odd, round towards zero
20:01:29speachyperhaps the most correct-est solution here is to just get rid of it altogether.
20:01:50speachybraewoods: well, it's really "truncate vs round" in this case, but yeah.
20:02:04braewoodsmaybe we could try using code from libsamplerate?
20:02:10braewoodsiirc, it was designed for this problem.
20:02:32braewoodsassuming the problem is scaling sample rates to fit the hardware rate
20:02:34dconradin terms of clicking/artifacting, not rounding at all seems to be the best solution, but maybe it's not the best for sound quality, I'm not sure
20:03:21speachyI'd think that "clicking/popping" is part of "sound quality" :D
20:03:28dconradwell, yeah... haha
20:04:02speachyI think that, if we don't round, we might never actually use +-MAX_INT
20:04:21speachywait wait.
20:04:46speachyif our fractional value is at MAX_U32, adding .5*UNITY will overflow
20:05:33speachyand since we shift, we'll end up with 0 instead of MAX_S16
20:06:07speachy(or more accurately, if our frac value is > (MAX_S32 - UNITY/2)
20:07:14speachythe more I think about it, the more it seems that we shouldn't do any rounding.
20:08:54speachybraewoods: does libsamplerate also deal with bitdepths too?
20:09:02speachybut it does seem to be fundamentally floating-point based.
20:09:06braewoodslet me look.
20:09:18braewoodsit was designed for desktop use, just the best example i could recall.
20:09:43dconradI don't quite follow why adding PCM_FACTOR_UNITY/2 is considered rounding tbh, but... wouldn't you want to look at the msb or 2 that get shifted out of the end value and round based on that?
20:09:52dconradmaybe that's what it is doing
20:10:18dconradlike rounding a 0.5 up or 0.4 down
20:10:27speachyin the scaling code, PCM_FACTOR_UNITY is basically equivalent to a sample value of "1"
20:11:13speachyso by using UNITY/2 you're adding the equivalent of 0.5, and then doing the divide/shift.
20:11:37speachy(you need to do it before the shift)
20:12:19braewoodsspeachy: seems it does rely on FP a lot
20:12:33braewoodsok, nevermind then
20:12:55braewoodsi just remembered it from all the packaging i did
20:13:28speachyfp-based is great for desktop applications (where you have fast FPUs even before you consider SIMD engines)
20:13:47braewoodsyea but most of our targets lack a FPU
20:13:56dconradso it sounds like that's basically what it is doing, but in a wacky bitwise way
20:14:09speachybut converting a fp algorithm to something integer-based is still as much of an art as a science.
20:15:17speachy(last two $dayjobs involved needing to convert multi-dimentional FP transforms into integer so they could be implemented in silicon)
20:16:05speachy(I wasn't doing that work but when your power budget is a couple of milliwatts, optimization _really_ matters)
20:16:11dconrad... though the device I'm interested in (erosq) does appear to have a fpu, would it work/be worth the effort to implement?
20:16:41dconrador "no way we have the cpu resources either way"
20:16:57speachyif we were using the linux port, sure, we get FP for free, but our bare-metal OS isn't FP-aware
20:17:07dconradah, I see
20:17:13speachy(we need to store/restore the FP registers as part of task switching)
20:17:39braewoodsspeachy: could we leverage the GCC fixed point extensions to help out?
20:17:54braewoodsit allows you to interact with them normally
20:18:04braewoodscompiler handles all the fixed point crapola so you don't need to simulate t
20:18:19braewoodsnot sure if 4.9 is new enough for that
20:18:23speachyI actually started implementing that a while back but given how expensive interrupt handling already is on our ingenic targts, throwing FP into the mix would probably render our audio playback unusable without further rewrites
20:18:56speachyversion 2 of that attempt was hacking in just enough FP support so we could use FP codecs.
20:18:58braewoodsi thought fixed point just used existing integer registers?
20:19:22speachywith the thinking that we'd never have more than one codec active so there wouldn't be any need to store/restore FPU state
20:19:54speachybraewoods: FP support, not gcc extensions
20:20:12braewoodsjust an idea i had if we wanted to revamp some code
20:21:09speachypretty sure gcc 4.9.x supports it, yes
20:21:48speachybut who knows how efficient it is.
20:22:12speachyoverflow/rounding isn't implemented anyway. :D
20:22:36braewoodsso we'd only get basic support
20:23:50braewoodsi just know it's a viable substitute for some uses of floating point
20:24:53braewoodsFP is too slow to be practical without the hardware support
20:25:04dconrad(if we're revamping code, my vote goes to >16bit volume scaling −−> dac tho, haha)
20:25:30braewoodsit won't be me; i don't know enough about audio signal stuff to write good code.
20:25:44speachydconrad: amen. start at the PCM driver API and work backwards from there
20:26:03braewoodsi'm currently busy with my inflate project anyway
20:26:11dconradit's ungodly complicated, I def understand why nobody's done it yet
20:26:39speachyfor sanity's sake we'd probably want to always go with whatever the DAC supports, rather than being able to switch between (eg) 16/24bpp.
20:26:46speachyon the fly, I mean
20:27:16braewoodswe could create an abstraction that exposes the audio IO and their capabilities
20:27:26braewoods(i don't know what rockbox currently does so yea
20:27:28speachywe already have that.
20:27:47braewoodsok good stuff
20:27:48speachyit's just that bitdepth is not in the set
20:28:03braewoodsbecause it was always hard coded before?
20:28:42speachyhigh-bitdepth stuff didn't hit the mainstream untill well after the general DAP market had long since been gutted by smartphones
20:28:53speachy(ie well after rockbox's heydey)
20:30:20dconradif we had a higher depth signal path all the way through, that might actually bring some audiophile types in as users I would think, they love that stuff
20:30:23speachybut anyway. starting at the PCM DAC API, working backwards, lets us do the minimal work to allow for softvol to work without touching most of the core
20:30:39speachyand we can push back incrementally from there
20:31:23speachybut other than FLAC, I don't think any of the codecs would ever care about high-res audio.
20:31:24dconradI was looking at that this morning, are you talking about like pcm_play_dma_status_callback_int and pcm_play_dma_complete_callback?
20:31:48dconradI think that's where the driver interfaces with the generic pcm code
20:32:01braewoodsspeachy: musepack supports up to 32 bit samples, iirc
20:32:02speachydconrad: that code wouldn't need to change actually −− but the code that fills those buffers would care.
20:32:10dconradI see
20:32:44speachyI think for sanity's sake we'd always want to send high-res samples to the DAC.
20:33:16speachyeg always operate the erosq at 24bit because making it runtime switchable greatly complicates the implementation
20:33:51dconradyeah I don't see any real reason to ever want less resolution, power saving maybe
20:34:25speachypower&|performance, but I don't think any of the targets where we're performance limited support high bitdepths anyway
20:35:28speachyafter the minimal work to enable softvol, the next step would be to push bitdepth awareness into the DSP core.
20:35:37dconradsupposedly we're probably only 2 bits short as it is, but apparently it makes a difference based on a noticeable difference between hosted and native
20:35:49speachyfrom there, individual PCM sample sources could be ported/enhanced as needed/convenient
20:36:20*speachy pages MrSomebody.
20:36:38dconradthat might go unanswered haha
20:37:16dconradbut I think I might give it another shot to do the swvol
20:37:27speachythe onlyuser of the "native" PCM API now is the plugin API
20:37:47speachy(well, other than our DSP layer)
20:38:04speachytbh we should move all of those users onto the DSP API instead.
20:44:22 Join massiveH [0] (
20:45:50dconradspeachy: so all that said, what's your thoughts on g#3633?
20:45:52rb-bluebotGerrit review #3633 at : ErosQ Native: Add DC Offset to PCM data by Dana Conrad
20:46:57dconradshould I go ahead and remove the preexisting rounding? I don't want to break any other targets...
20:47:14speachyIMO the proper solution is to nuke the rounding. it's clearly broken.
20:47:31dconradI can do that
20:47:40speachythe DC offset shouldn't be needed if the rounding is gone.
20:48:01dconradI do believe it will click without it, I'll try real quick
20:50:31dconradyeah, it clicks on play/pause and going in/out of mute
20:50:39dconradplayback is clean though
20:50:55dconradI think the clicking is a hardware-specific bug
20:51:29speachydoes that include tha change here:
20:51:58dconradyeah, that's the muting
20:52:36speachyis INT_MIN negative, or 0?
20:52:55speachyfrom the dac's pecspective, mute == 0
20:53:28dconradlet me try to find that
20:53:28speachy(since we're talking about a signed sample)
20:54:09speachy(PCM_MUTE_LEVEL needs to be 0)
20:54:46dconradI think PCM_MUTE_LEVEL is really just a signal to tell pcm_centibels_to_factor() to return 0 regardless
20:55:03dconradwhich with the DC offset will be dc -1
20:55:26speachydc -1 will prevent the DAC from auto-muting.
20:55:39dconradthat's probably why it works to keep it from clicking
20:55:52dconradthough I don't know why +1 doesn't do the same thing
20:56:33speachyso what I suggest is twofold, nuke the rounding first as that's clearly jacked. and in the erosq code, don't specify PCM_MUTE_LEVEL and instead use an explicit -1 or whatever instead.
20:58:14dconradI can do the first, but the second, how can I ensure that except to go through pcm_set_master_volume()? I don't quite follow
20:58:48speachyI mean, in audiohw_set_volume() in eros_qn_codec.c
20:59:15speachyl = l << PCM5102A_VOLUME_MIN ? -1 : l;
20:59:25speachyoh wait nevermind
21:00:08dconradyeah, that's gonna get mangled into a multiplicative factor unless it's PCM_MUTE_LEVEL
21:00:27speachy"muted" means "still playing but at 0 volume"
21:00:54dconradI did try using the hardware muting stuff but everything clicks at least a little
21:01:44speachyhmm. I wonder what the XSMT pin on the codec is doing.
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21:01:50dconradwhich is really funny, all the datasheets say "this part is POP AND CLICKLESS" and like... no it's not haha
21:02:06dconradyeah, I tried that earlier - it works, but it clicks
21:02:17speachy"when the device detects continuous zero data, it enters a full analog mute condition"
21:02:33dconradI can give it a try with the DC offset, I don't know if I tried that
21:02:58dconrador just try it again without just to double check
21:03:11speachyyou are controlling that pin?
21:04:23dconradwell, there's auto-mute and the XMIT pin, I think they're separate
21:04:42dconradauto-mute, I think means it detected that the clock stopped
21:05:00speachyclock stopped OR the sample data is all zeros for a certian amount of time
21:05:11 Join munkis [0] (
21:05:15dconradaha, that's probably the fundamental issue right there
21:05:24speachythat's why the DC offset works
21:05:31speachyprevents the codec from muting.
21:05:46dconradthough I do wonder why -1 works and +1 doesn't, magic probably
21:05:54speachybtw, the +1 doesn't work there because it's shifted away, whereas the negative value remains negative due to the lack of rounding
21:06:04dconradoh, well there ya go
21:07:26dconradwould you want removing the rounding to be its own commit?
21:07:28speachyso perhaps instead, in pcm_hw_set_vol() you can assert the XSMT instead?
21:07:38speachyif the volume is min, I mean
21:07:54dconradI can try, but unfortunately do think it's going to be audible
21:08:41speachyand yes, remove rounding in a separate commit, or I can do it instead.
21:08:48dconradwait, instead of the DC offset you mean? or the PCM_MUTE_LEVEL, or both
21:09:00speachyyes, instead of both
21:09:10dconradgotcha, I'll try it
21:09:18speachy(with the rounding removed)
21:14:48dconradok, so with rounding removed, no DC bias, and using the XMIT pin as muting, we have:
21:15:44dconradplayback is clean, muting/unmuting clicks, any start/stop/play/pause or any operation that starts/stops pcm data clicks
21:15:59*speachy frowns.
21:16:33speachyokay, what happens if you toggle the XSMT when volume is set to "normal" levels?
21:16:51dconradhm, I'm not sure how to accomplish that
21:17:22dconradany easy places to plop it in come to mind?
21:17:23speachyhack something into the button code
21:17:29speachyeg press M for mute. :D
21:17:51speachy(ie in the erosq's button code, before it gets sent to the rockbox core)
21:18:03dconradah, I'll give it a try
21:18:05speachyI used the lock switch on the X3 to accomplish similar stuff
21:22:14speachybraewoods: btw, this was my WIP FPU code: g#2814
21:22:16rb-bluebotGerrit review #2814 at : WIP: Native MIPS supports FPU. (DO NOT MERGE) by Solomon Peachy
21:24:11dconradok, so I used vol+/- to mute/unmute at -45dB
21:24:47dconradyou can still hear the click a little, though it doesn't seem to scale up with volume so it's buried a little
21:24:55dconradthough I'm not sure what that proves tbh
21:25:10speachyit proves it's a hw artifact.
21:26:19speachyoh, are you using HP or LO out?
21:26:41dconradI suppose I can go try LO out, I think I tried that earlier as well
21:26:49speachythe OF disables the HP and/or LO output altogether
21:27:09speachyif you mute, you switch outputs to something that's not connected
21:27:15speachythen you don't get a pop
21:27:34speachyfrom the codec being special
21:27:52dconradhm, I'm pretty sure I ohm'd out the connections on the LO jack direct to the dac
21:28:39speachymight be worth trying this via LO instead of HP, to eliminate that as the source of the click
21:28:52dconradyeah I can try that
21:28:55dconradone second
21:29:54speachythe HP amp presumably has a mute too?
21:31:22dconradok, back
21:32:10dconradthe click on play/pause/mute are all there on the LO out, though I had to turn the volume way down manually to hear it clearly, no surprise there
21:32:14dconradbut, it's present
21:32:22dconradyeah the HP amp has a mute as well
21:32:32dconradI as does the stereo switch
21:32:53speachyso mute the switch? :D
21:33:16speachy(and the HP amp should be muted too, if there are no headphones inserted..)
21:33:49dconradin terms of least to most loud click, it goes HP power = switch mute = dac xmit = switch input sel, then hp mute, then dac analog power
21:34:09dconrad(I tested all those earlier)
21:34:18dconradbut... all click
21:34:34dconrad(at least a little)
21:36:49dconradthe dac analog power is a proper pop, which I guess shouldn't be surprising
21:38:10dconradI suppose I can test the hp amp and switch mutes again just for fun
21:38:20dconradwhat with rounding removed
21:39:06dconradthough that wouldn't remove the pcm data start/stop clicking, especially when using the LO
21:39:33*speachy nods.
21:40:05speachywell, adding in HP muting is a good idea regardless (if only to save power when no HP is present)
21:41:03dconradyeah I can see what happens if I power down the HP amp when headphones are detected as removed
21:41:16dconradhopefully it doesn't cause an audible artifact on the LO
21:41:42dconrador set it to mute, since there won't be anything to hear its click anyway
21:47:36dconradok, so that got messy but I think I cleaned it up
21:48:21dconradso I'm going to try this with DC offset, with PCM_MUTE_LEVEL (though that shouldn't affect anything)
21:48:44dconradand no rounding obvs
21:55:41dconradok, there don't seem to be any weird side effects of muting the headphone amp when the headphones are inserted/removed
21:56:02dconradthe LO seems to be completely unaffected
22:10:29dconrad g#3635, simple enough
22:10:31rb-bluebotGerrit review #3635 at : Software Volume: Remove sample rounding by Dana Conrad
22:11:01dconradand now that we've eliminated all other options, g#3633 is updated
22:11:03rb-bluebotGerrit review #3633 at : ErosQ Native: Add DC Offset to PCM data by Dana Conrad
22:18:26rb-bluebotBuild Server message: New build round started. Revision ecf0d631e7, 303 builds, 9 clients.
22:20:13speachydconrad: you didn't update 3633
22:20:38dconrad?? oh, I will need to fix the merge conflict
22:20:50dconradwon't take but a second
22:24:15dconradok, merge conflict resolved
22:24:47 Quit cockroach (Ping timeout: 252 seconds)
22:29:31rb-bluebotBuild Server message: Build round completed after 665 seconds.
22:29:34rb-bluebotBuild Server message: Revision ecf0d631e7 result: All green
22:29:38rb-bluebotBuild Server message: New build round started. Revision a2fd00cf38, 303 builds, 9 clients.
22:39:44rb-bluebotBuild Server message: Build round completed after 606 seconds.
22:39:46rb-bluebotBuild Server message: Revision a2fd00cf38 result: All green
23:01:10 Quit skipwich (Read error: Connection reset by peer)
23:01:51***Saving seen data "./dancer.seen"
23:54:01 Quit dconrad (Remote host closed the connection)

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