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#rockbox log for 2017-10-02

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04:42:07fs-bluebotBuild Server message: 3New build round started. Revision b2a373e, 273 builds, 13 clients.
04:50:07fs-bluebotBuild Server message: 3Build round completed after 480 seconds.
04:50:08fs-bluebotBuild Server message: 3Revision b2a373e result: 340 errors 0 warnings
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05:28:32fs-bluebotBuild Server message: 3New build round started. Revision 66b49dc, 273 builds, 13 clients.
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05:44:14fs-bluebotBuild Server message: 3Build round completed after 941 seconds.
05:44:15fs-bluebotBuild Server message: 3Revision 66b49dc result: All green
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19:58:13lebelliumpamaury: can the frequency be set automatically according to the contents?
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20:39:03pamaurylebellium: no
20:39:17pamaurywe don't support that
20:45:21lebelliumwouldn't it make more sense than setting a fixed frequency?
20:50:00jhMikeSno, because changing the hardware frequency is glitchy and has many side-effects
20:51:41lebelliumhow do all modern 'audiophile' OF achieve that then?
20:54:33jhMikeSassuming they're really achieving what one might assume
20:55:20jhMikeSthe hardware design matters. the features that are expected to work over a sample rate change matter.
20:55:41gevaertsOne way to do it is to set the actual frequency to the lowest common multiple of the supported frequencies :)
20:55:46jhMikeSthe dsp and hardware do have histories of samples that will have to be discarded
20:56:03*jhMikeS slaps gevearts
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21:00
21:00:03pamauryjhMikeS: by the way, on the sony A15, the dac supports crazy high frequencies, I suspect it's only a matter of time before people ask we can support then. What is required to add a frequency to the list?
21:02:29lebelliumpeople probably won't ask to support it if they have to manually switch between 44,1 Khz and 192 Khz between each song... And I don't think there are many people having all their files with the same frequency rate (if different from 44,1)
21:03:09jhMikeSpamaury: I suppose define more frequency cap bits
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21:05:44jhMikeSpamaury: firmware/export/pcm_sampr.h is where that stuff is. check code for any assumptions.
21:12:31saratogai think some devices already do 96k support (but not for playback)
21:12:56pamaurylebellium: one can just set to 192KHz permanently, no need to switch for each track
21:13:18saratogaright now, we don't allow that, and i suspect it would do bad things to some of the dsp code
21:13:41pamaurysaratoga: what do we not allow?
21:13:51saratoga>48k at playback
21:14:09pamauryah, what is missing? some dsp code assume 44.1 or 48?
21:14:21pamaury(that was my question about new sample rates)
21:14:40saratogaprobably buffers will be too small in places if you go to very high sample rates
21:15:02saratogabut mostly it is about efficiency and discouraging people from doing dumb stuff like just maxing out the sample rate
21:15:27saratogai think the only places where 96khz is used is in some plugins
21:16:45pamaurypersonally I think 44.1 is enough for everyone, maybe 48 but above is stupid
21:16:55lebelliumpamaury: I'm not an expert but isn't playing 44,1 Khz files at 44,1 Khz better than playing them at 192 KHz (conversion)?
21:17:44saratogathe new resampler is pretty good, so i doubt you notice much difference in practice, but it is certainly much more efficient to use lower sampling rates
21:17:55saratogaefficient as in power consumption
21:18:13jhMikeSsaratoga: the DSP does reserve enough for the maximum allowed where it matters, like in crossfeed
21:18:30pamaurysaratoga: by the way, I'm looking into adding support for 24-bit and 32-bit sample size in rockbox (it is needed for the Sony, because Sony is stupid). I can easily add support for a new sample size in the dsp code itself (because it does everything in 32-bit internally). But do you think the pcm mixer code assumes 16-bit?
21:18:58pamauryalso I don't know what to do about plugins
21:19:01lebelliumThe thing is that the selling argument of all those new devices is the native playback of "HD" files so supporting them in Rockbox may help to get some people interested in rockboxing their device even though it's not a commercial product and we don't want to sell something
21:19:18jhMikeSpamaury: it all assumes 16 bit. I was going to work on that buy I need to know if it was a compile-time setting or changed at runtime
21:19:19pamauryif suddenly the raw pcm interface expects 32-bit sample size, virtually all plugins with audio will break
21:20:10pamauryjhMikeS: it's safe to say it's compile time. Sony's driver itself accepts 16-bit and 32-bit but 16-bit is basically broken so I'll drive in 32-bit all the time
21:20:58pamauryso my question about plugins is: 1) do we keep a 16-bit raw interace that does conversion behind the scenes? 2) or do we fix all plugins to do the conversion before?
21:22:26jhMikeSpersonally, I was leaning towards 1) being the default
21:23:46pamaurythat means we'll have to have some buffer somewhere? I guess we can safely split the input buffer if needed so it only needs to be big enough to avoid underflow, is that what is done in the dsp code too?
21:24:47jhMikeSthe mixer already has a buffer and changing depth can just be part of the mixdown
21:25:35pamauryare the samples always going through the mixer? even for plugins?
21:26:07jhMikeSno. for other cases, do it in your hardware driver to up it?
21:26:52pamauryso we'll have to pcm interfaces, one for 16-bit, one for native? (otherwis driver cannot know)
21:26:56pamaury*two pcm
21:26:58jhMikeSonly mixer based plugin is mpegplayer afaik
21:27:39 Quit Bilgus (Ping timeout: 248 seconds)
21:28:30jhMikeSit didn't appear you need two separate ones, just tell it what you're using when starting playback. maybe a pcm_play_start_ex that specifies a depth
21:30:16pamaurygood idea
21:30:30pamauryso are you going to work on this? I don't want to duplicate the work
21:34:57jhMikeSI did start it already. I was just wondering about whether DSP would always be 32-bit.
21:35:15pamauryisn't the DSP always 32-bit?
21:35:31pamauryalso wodz mentioned he is interested in a 24-bit sample size
21:35:44jhMikeSinternally, but the dsp output functions do the truncation to 16
21:35:54pamauryright
21:36:17jhMikeSI suppose I don't want 24-bit packed. that would be slow as hell
21:36:56jhMikeSbesides, no hardware accepts 24 bit as packed anyway
21:37:07pamauryI guess any decent pcm dma would be able to do unpacked 24-bit
21:37:29pamauryI don't remember if i2s does 24-bit packed or not
21:37:48jhMikeSno, since the FIFO registers are 32-bit
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21:38:45jhMikeSthe same modifiied code would be good for anything from 17 to 32 bit
21:40:03pamaurythe ATJ has an internal DAC that takes 24-bit sample, it's not clear from the datasheet if it can take 32-bit
21:42:18jhMikeSI think I2S just ignores least significant bits beyond the maximum precision
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21:47:44jhMikeSI guess I'm going to need to pick a player and modify the PCM to handle more than 16 bits. probably gigbeat-s
21:48:49pamauryyou mean for testing? you can do a simulator build and add 32-bit pcm alsa support :-p
21:51:06jhMikeSyeah, but I need integrate things for real hardware, whatever comes up
21:52:14pamauryare for the optimized assembly routines?
21:54:39jhMikeSnothing to do with that, but there will need to be more of those
21:56:08pamaurywhat is needed and requires a real target then?
21:58:04jhMikeSnot sure at the moment tbh
21:59:12jhMikeSpossibly the part where the driver accepts 16-bit input
22:00
22:17:19__builtinjhMikeS: g#1674 isn't meant to be pushed or anything
22:17:21fs-bluebot3Gerrit review #1674 at http://gerrit.rockbox.org/r/1674 : 3Action.c FIX / Rework / Clean-up Action System by William Wilgus
22:17:47__builtinG#1679 i mean
22:17:49fs-bluebot3Gerrit review #1679 at http://gerrit.rockbox.org/r/1679 : 3[HACK] Support floating-point formatting by Franklin Wei
22:20:19jhMikeS__builtin: okay :)
22:46:06fs-bluebotBuild Server message: 3New build round started. Revision 3f1e4a0, 273 builds, 14 clients.
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22:56:12fs-bluebotBuild Server message: 3Build round completed after 606 seconds.
22:56:13fs-bluebotBuild Server message: 3Revision 3f1e4a0 result: All green
22:59:25AldemRIP Tom Petty :/
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