Alsa-devel August 2018

alsa-devel@alsa-project.org

109 participants
267 discussions

[alsa-devel] [PATCH 0/6] ASoC: rt5651: Various fixes and quirks
by Hans de Goede 24 Dec '18

24 Dec '18

Hi All, Here is my last (big) round of (Intel) rt5651 support updates, with these patches combined with the updated, matching UCM profiles from here: https://github.com/jwrdegoede/alsa-lib/commits/master All 7 different devices I have work 100% OOTB, without requiring any manual config or mixer setup. After this there should be no more big changes. I will submit the matching alsa-lib UCM profile changes upstream once this set is in -next (and thus I can be sure the long-names will not change due to review). Note that unfortunately my previous round of cleanup patches removed 1 input mapping (of the 3 mappings in total it removed) which is actually necessary. So this series re-introduces that 1 mapping. I did not notice this before because UCM input profile switching does not work reliably for some reason, something which I still need to investigate. This time I manually veryfied that the correct / expected mixer settings where used while testing. Regards, Hans

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[alsa-devel] [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by Hans de Goede 28 Nov '18

28 Nov '18

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is identical to all the cset statements this commit removes. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/chtrt5645/HiFi.conf | 116 +------------------------------ 1 file changed, 1 insertion(+), 115 deletions(-) diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index e81866cf..d993f6ae 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,121 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - # media mixer settings - # compress - cset "name='media0_in Gain 0 Switch' on" - cset "name='media0_in Gain 0 Volume' 0" - - # normal - cset "name='media1_in Gain 0 Switch' on" - cset "name='media1_in Gain 0 Volume' 0" - # swm loopback - cset "name='media2_in Gain 0 Switch' off" - cset "name='media2_in Gain 0 Volume' 0%" - # deep buffer - cset "name='media3_in Gain 0 Switch' on" - cset "name='media3_in Gain 0 Volume' 0" - - cset "name='media0_out mix 0 media0_in Switch' on" - cset "name='media0_out mix 0 media1_in Switch' on" - cset "name='media0_out mix 0 media2_in Switch' off" - cset "name='media0_out mix 0 media3_in Switch' on" - - cset "name='media1_out mix 0 media0_in Switch' off" - cset "name='media1_out mix 0 media1_in Switch' off" - cset "name='media1_out mix 0 media2_in Switch' off" - cset "name='media1_out mix 0 media3_in Switch' off" - - cset "name='pcm0_in Gain 0 Switch' on" - cset "name='pcm0_in Gain 0 Volume' 0" - - cset "name='pcm1_in Gain 0 Switch' off" - cset "name='pcm1_in Gain 0 Volume' 0%" - - # codec0_out settings (used if ssp2 is connected to aif1) - cset "name='codec_out0 mix 0 codec_in0 Switch' off" - cset "name='codec_out0 mix 0 codec_in1 Switch' off" - cset "name='codec_out0 mix 0 media_loop1_in Switch' off" - cset "name='codec_out0 mix 0 media_loop2_in Switch' off" - cset "name='codec_out0 mix 0 pcm0_in Switch' on" - cset "name='codec_out0 mix 0 pcm1_in Switch' off" - cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out0 Gain 0 Switch' on" - cset "name='codec_out0 Gain 0 Volume' 0" - - # modem_out settings (used if ssp0 is connected to aif2) - cset "name='modem_out mix 0 codec_in0 Switch' off" - cset "name='modem_out mix 0 codec_in1 Switch' off" - cset "name='modem_out mix 0 media_loop1_in Switch' off" - cset "name='modem_out mix 0 media_loop2_in Switch' off" - cset "name='modem_out mix 0 pcm0_in Switch' on" - cset "name='modem_out mix 0 pcm1_in Switch' off" - cset "name='modem_out mix 0 sprot_loop_in Switch' off" - cset "name='modem_out Gain 0 Switch' on" - cset "name='modem_out Gain 0 Volume' 0" - - # input settings - # pcm1_out settings - - # input used when SSP2 is connected - cset "name='codec_in0 Gain 0 Switch' on" - cset "name='codec_in0 Gain 0 Volume' 0" - - # input used when SSP0 is connected - cset "name='modem_in Gain 0 Switch' on" - cset "name='modem_in Gain 0 Volume' 0" - - cset "name='pcm1_out mix 0 codec_in0 Switch' on" - cset "name='pcm1_out mix 0 modem_in Switch' on" - cset "name='pcm1_out mix 0 codec_in1 Switch' off" - cset "name='pcm1_out mix 0 media_loop1_in Switch' off" - cset "name='pcm1_out mix 0 media_loop2_in Switch' off" - cset "name='pcm1_out mix 0 pcm0_in Switch' off" - cset "name='pcm1_out mix 0 pcm1_in Switch' off" - cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" - - cset "name='pcm1_out Gain 0 Switch' on" - cset "name='pcm1_out Gain 0 Volume' 0" - - # disable codec_out1 - cset "name='codec_out1 mix 0 codec_in0 Switch' off" - cset "name='codec_out1 mix 0 codec_in1 Switch' off" - cset "name='codec_out1 mix 0 media_loop1_in Switch' off" - cset "name='codec_out1 mix 0 media_loop2_in Switch' off" - cset "name='codec_out1 mix 0 pcm0_in Switch' off" - cset "name='codec_out1 mix 0 pcm1_in Switch' off" - cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" - cset "name='codec_out1 Gain 0 Switch' off" - cset "name='codec_out1 Gain 0 Volume' 0%" - - # disable codec_in1 - cset "name='codec_in1 Gain 0 Switch' off" - cset "name='codec_in1 Gain 0 Volume' 0%" - - # disable all loops - cset "name='media_loop1_out mix 0 codec_in0 Switch' off" - cset "name='media_loop1_out mix 0 codec_in1 Switch' off" - cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop1_out mix 0 pcm0_in Switch' off" - cset "name='media_loop1_out mix 0 pcm1_in Switch' off" - cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" - - cset "name='media_loop2_out mix 0 codec_in0 Switch' off" - cset "name='media_loop2_out mix 0 codec_in1 Switch' off" - cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" - cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" - cset "name='media_loop2_out mix 0 pcm0_in Switch' off" - cset "name='media_loop2_out mix 0 pcm1_in Switch' off" - cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" - - cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" - cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" - cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" - cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" - cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" - cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + <bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" -- 2.18.0

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Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 27 Nov '18

27 Nov '18

Please don't drop Cc to ML. Also avoid top-posting. On Sat, 04 Aug 2018 19:02:53 +0200, David Ulricht wrote: > > How can I do that ? > I tried verifying if /proc/asound/card0/codec#0 changes when i put > handphone jack in and out with "diff ver1 ver2" and it is exactly the same. Just do for each bit as you tried for GPIO output. In this case, the direction bit should be different (set 0), and you need to read. > > One strange thing I note is hda-jack-retask.fw contains: > [pincfg] > 0x24 0x90100080 > 0x25 0x90100082 > I have 0x90100080 for 0x24 but in /proc/asound/card0/codec#0 : > Node 0x24 [Pin Complex] wcaps 0x400101: Stereo > Pincap 0x00000010: OUT > Pin Default 0x90100110: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x02 > > Shouldn't "Pin Default 0x90100110" be 0x90100080 after applying the patch ? No, the codec config value itself won't change, the driver uses the given value only internally. Takashi > I can see that Applying patch firmware 'hda-jack-retask.fw' is before > hdaudioC0D0: autoconfig for Generic: line_outs=2 (0x24/0x25/0x0/0x0/0x0) > type:speaker > in dmesg and this worries me, maybe the pins are overriden by generic > cirrus logic ? > > > On Sat, Aug 4, 2018 at 7:26 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 17:44:59 +0200, > > David Ulricht wrote: > > > > > > I used the following script running as root to verify the 8 GPIOs while > > > playing long youtube video, > > > I have running pulseaudio meanwhile and not restarting it, i think its > > okay > > > like that people on the ubuntu forums say that after using > > > hda-verb and setting a gpio active immediately sound appears, so i guess > > > there is no need to restart pulseaudio. > > > I have done the same test with headphones in. > > > No sound in both scenarios, no input no output detected (i'm watching > > > pavucontrol for input detection). > > > > This codec is fairly unique and doesn't provide the standard jack > > detection on each pin. Maybe it's via either GPIO or any other > > method. You can try to read GPIO pins (set as input) to see whether > > any of them corresponds to the jack detection, for example. > > > > > > Takashi > > > > > > > > #!/usr/bin/env python3 > > > import os,time > > > for x in range(0,256): > > > mhex=("0x%0.2X" % x) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > > > time.sleep(3) > > > > > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > > > IO[0-7] changes its configuration, > > > in comparison Joel Krahemann in his scripts is changing the hex right > > after > > > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > > > correctly. > > > > > > I found the following post: > > > https://forum.manjaro.org/t/sound-not-working-on-a-2017- > > imac-27-5k-retina/43638 > > > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 > > has > > > a similar soundcard. > > > How can i setup with command line or any other utility: > > > """select “off” in the top menu that propose to setup the Digital > > > Surround""" > > > this is some setting in xfce4-mixer package which is not available for > > > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > > > find this setting in audio mixer of xfce4 either. > > > > > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > > > David Ulricht wrote: > > > > > > > > > > Hello, > > > > > > > > > > Macbook pro models containing CS8409 for sure are: > > > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > > > > > I'm attaching working sound configuration from Windows 10 registry > > (note > > > > > that the cs420x might need to be ignored or might be important i > > don't > > > > > really know). I believe what is really used is in the > > PinConfigOverride > > > > > section. > > > > > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > > > and your override (user_pin_cfg), there aren't so many changes. > > > > All changes are pretty minor, and I guess it won't influence on the > > > > driver behavior. > > > > > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > > > Or it's some direct I2C control. The INI file mentions it. > > > > And that's above HD-audio driver's responsibility. > > > > > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > > > > > It's because the codec chip has no output amplifier at all. > > > > So there can be neither output volume nor mute control on this chip. > > > > > > > > > What I have tried is execute: > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > > > putting @@ from 0x00 to 0x50 > > > > > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > > > 0x02, 0x04, 0x08, ... 0x80. > > > > > > > > And how is the current situation? You can't play *and* record > > > > anything from any inputs / outputs? > > > > > > > > > > > > Takashi > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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Re: [alsa-devel] [linux-sunxi] [BUG] On Allwinner A20 Board Cubietruck distorted SPDIF opical output
by Markus Mitsch 27 Oct '18

27 Oct '18

hi. i posted my answer wrong so here is my finding: i made the output work by disabling the sigma delta modulation. after compiling the kernel i can output through spdif with any rate. greetings markus 2018-05-26 18:38 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: > 2018-05-26 16:57 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >> 2018-05-23 17:12 GMT+02:00 Code Kipper <codekipper(a)gmail.com>: >>> Hi All, >>> I can see that the problem was introduced somewhere here >>> >>> DOESN'T WORK >>> ffc3eb6f3a83 Merge tag 'sunxi-clk-for-4.15' of >>> https://git.kernel.org/pub/scm/linux/kernel/git/sunxi/linux into >>> clk-next >>> ae74ac082886 Merge tag 'clk-v4.15-exynos-pm' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> faa865f18cb7 Merge tag 'clk-v4.15-samsung' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> f09a6b86fdae Merge tag 'clk-renesas-for-v4.15-tag1' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/geert/renesas-drivers >>> into clk-next >>> 4328a2186e51 clk: sunxi-ng: sun4i: Export video PLLs >>> 553c7d5ba2fe clk: sunxi-ng: Add A83T display clocks >>> 7679eb20353d clk: samsung: Add a separate driver for Exynos4412 ISP clocks >>> 8ca8ac102484 clk: samsung: Add dt bindings for Exynos4412 ISP clock controller >>> 75920aac275a clk: samsung: Instantiate Exynos4412 ISP clocks only when available >>> 24ea78a09f89 clk: sunxi-ng: sun8i: a23: Use sigma-delta modulation for audio PLL >>> ee6501d69217 clk: sunxi-ng: sun6i: Use sigma-delta modulation for audio PLL >>> 042f7f8f9715 clk: sunxi-ng: sun5i: Use sigma-delta modulation for audio PLL >>> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >>> a5e3e2b2ef85 clk: sunxi-ng: sun8i: h3: Use sigma-delta modulation for audio PLL >>> 392ba5fafcdf clk: sunxi-ng: nm: Add support for sigma-delta modulation >>> 05d2eaac96d4 clk: sunxi-ng: Add sigma-delta modulation support >>> 4cdbc40d64d4 clk: sunxi-ng: nm: Check if requested rate is supported >>> by fractional clock >>> d51fe3ba9773 clk: sunxi-ng: sun5i: Fix bit offset of audio PLL post-divider >>> 4e8975cbb516 clk: samsung: exynos5433: mark PM functions as __maybe_unused >>> a4f21e9ceb5c clk: samsung: Remove obsolete clkdev alias support >>> 45d882daf88a clk: samsung: Add explicit MPLL, EPLL clkdev aliases in >>> S3C2443 driver >>> cd05417f728b clk: samsung: Rework clkdev alias handling in S3C2443 driver >>> efea8d377184 clk: samsung: Rework clkdev alias handling in Exynos5440 driver >>> 36ba48240b19 clk: samsung: Drop useless alias in Exynos5420 clk driver >>> 29964890f31c clk: samsung: Remove clkdev alias support in Exynos5250 clk driver >>> 6de08891c896 clk: samsung: Remove double assignment of CLK_ARM_CLK in >>> Exynos4 driver >>> 58f4a5ff3a62 clk: samsung: Remove clkdev alias support in Exynos4 clk driver >>> c9194fb623b0 clk: samsung: Remove support for obsolete Exynos4212 CPU clock >>> d5cd103b06f9 clk: samsung: Remove support for Exynos4212 SoCs in >>> Exynos CLKOUT driver >>> 7c4f63ec94a1 clk: sunxi-ng: a83t: Fix invalid csi-mclk mux offset >>> WORKS >>> 0a4e632b6f9d Merge branch 'clk-fixes' into clk-next >>> >>> BR, >>> CK >>> >>> >>> On 22 May 2018 at 21:27, Code Kipper <codekipper(a)gmail.com> wrote: >>>> Hi Markus, >>>> I've been able to reproduce this on my A31 board with linux-next(plays >>>> at 44100 but not at 96K)so it is an issue. I also tested the same code >>>> on my A64 and it worked fine, The blocks are very similar so it maybe >>>> a clocking issue, I'll look into this. >>>> BR, >>>> CK >>>> >>>> On 21 May 2018 at 20:17, Markus Mitsch <markusmitsch86(a)googlemail.com> wrote: >>>>> 2018-05-21 15:17 GMT+02:00 Maxime Ripard <maxime.ripard(a)bootlin.com>: >>>>>> On Mon, May 21, 2018 at 02:43:55PM +0200, Markus Mitsch wrote: >>>>>>> 2018-05-21 11:43 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >>>>>>> >>> Hello, >>>>>>> >>> >>>>>>> >>> i have a new problem. if i want to output sound from optical i get >>>>>>> >>> distorted sound. At 44.1 khz its like an old vinly, but at 48 khz and >>>>>>> >>> higher it is very bad. thanks in advance for your time. >>>>>>> > >>>>>>> >> Hi Markus, >>>>>>> >> I haven't tested it for a while but when I did it worked as high as >>>>>>> >> 96K with no issues. Does it work ok with the original cubietruck >>>>>>> >> software?, >>>>>>> >> CK >>>>>>> > >>>>>>> > Hello, >>>>>>> > i am using arch linux with mainline kernel 4.16.8. parhaps i can try >>>>>>> > today with official image. >>>>>>> > mfg >>>>>>> >>>>>>> Hi. >>>>>>> So, with debian image from 2014 with 3.4.79 kernel the sound is clear >>>>>>> up to 96 khz. >>>>>> >>>>>> You said that it is a new problem, when was it introduced? Could you >>>>>> bisect the issue? >>>>>> >>>>>> Maxime >>>>> >>>>> Hello. >>>>> I dont know since when this problem appears. When saying "new" i >>>>> refered to my older post. I tried to look into the dts file and the >>>>> code >>>>> but i am afraid this is a bit too high for me. I can say that the >>>>> analog output works, even at high sampling rates. >>>>> And I have read something about a clock source could be the problem. >>>>> I'm sorry I can't tell you any more. >>>>> >>>>> mfg >>>>> Markus >> >> Hello, >> i tried undoing part of the commit >> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >> by setting the value for the divider from 1 to 4 again on both places. >> with no success. >> >> markus > > i now had a little success. the output works!! > i disabled the sigma-delta support by making > ccu_sdm_helper_enable in drivers/clk/sunxi-ng/ccu_sdm.c return imediately. > > markus

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Re: [alsa-devel] [PATCH 1/3] MIPS: jz4780: Allow access to jz4740-i2s
by Paul Burton 17 Oct '18

17 Oct '18

Hi Matthieu, On Wed, Jun 06, 2018 at 09:38:08PM +0200, Mathieu Malaterre wrote: > diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig > index 1a354a6b6e87..35d82d96e781 100644 > --- a/sound/soc/jz4740/Kconfig > +++ b/sound/soc/jz4740/Kconfig > @@ -1,20 +1,20 @@ > config SND_JZ4740_SOC > - tristate "SoC Audio for Ingenic JZ4740 SoC" > - depends on MACH_JZ4740 || COMPILE_TEST > + tristate "SoC Audio for Ingenic JZ4740/JZ4780 SoC" > + depends on MACH_JZ4740 || MACH_JZ4780 || COMPILE_TEST Perhaps this could be MACH_INGENIC, or even just MIPS? Thanks, Paul

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[alsa-devel] [ANNOUNCE] 2018 Linux Audio miniconference
by Mark Brown 16 Oct '18

16 Oct '18

Hi, As in previous years we're trying to organize an audio miniconference so we can get together and talk through issues, especially design decisons, face to face. This year's event will be held on Sunday October 21st in Edinburgh, the day before ELC Europe starts there. We're still finalizing the venue (we expect to be able to confirm next week) but it'll be outside the ELC venue so there will be no need to purchase an ELC ticket to attend. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/spreadsheets/d/11p2pSNxS7sJysg1JwErWi0T2Dq7XKRtAGng… Thanks, Mark

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[alsa-devel] [PATCH v2 00/12] Sound Open Firmware Core
by Liam Girdwood 20 Sep '18

20 Sep '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since V1 :- o Added PM source. o Updated header comments to C++ style o Used generic private data in snd_soc_pcm_runtime o remove sof_get_pages() o Added comments for compressed SG pages. o Check return values for runtime_pm calls. Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add PM support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 3 + include/sound/sof.h | 82 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 67 + include/uapi/sound/sof-ipc.h | 938 ++++++++++++ include/uapi/sound/sof-tone.h | 27 + include/uapi/sound/sof-topology.h | 94 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 36 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 270 ++++ sound/soc/sof/core.c | 381 +++++ sound/soc/sof/debug.c | 157 ++ sound/soc/sof/ipc.c | 788 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 ++++ sound/soc/sof/pcm.c | 733 ++++++++++ sound/soc/sof/pm.c | 345 +++++ sound/soc/sof/sof-priv.h | 519 +++++++ sound/soc/sof/topology.c | 2222 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 293 ++++ 24 files changed, 7889 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

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[alsa-devel] [RFC - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 12 Sep '18

12 Sep '18

Hi all, This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit/receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener. Details about the plugin implementation are provided in the patches description, and information on how to configure and run the AAF plugin is provided in doc/aaf.txt file. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1172 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 110 ++++++ 5 files changed, 1303 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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[alsa-devel] [PATCH - DMA engine v2 1/1] ALSA: SOC: DMA: increment buffer pointer atomically
by twischer＠de.adit-jv.com 11 Sep '18

11 Sep '18

From: Andreas Pape <apape(a)de.adit-jv.com> Setting pointer and afterwards check for wrap around leads to the possibility of returning the inconsistent pointer position. This patch increments buffer pointer atomically to avoid this issue. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Since commit: b7ae6f3 dmaengine implementation has been moved from ASoC to ALSA core. So change goes to sound/soc/soc-dmaengine-pcm.c, apply to sound/core/pcm_dmaengine.c now cherry picked from ADIT v3.14 e2f803f("ALSA: SOC: DMA: increment buffer pointer atomically") Signed-off-by: Veeraiyan Chidambaram <veeraiyan.chidambaram(a)in.bosch.com> (cherry picked from ADIT v4.1 commit 9e7eff10a38157db75f12e032fc378d3104a2812) Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> (cherry picked from ADIT v4.9 commit 66c00aa1e7cd3b077fc2e5a45a4594b86246cf78) Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- v1 can be found in http://mailman.alsa-project.org/pipermail/alsa-devel/2016-November/115277.h… diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 8eb58c7..0a7ed85 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -139,12 +139,14 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { + unsigned int new_pos; struct snd_pcm_substream *substream = arg; struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; + new_pos = READ_ONCE(prtd->pos) + snd_pcm_lib_period_bytes(substream); + if (new_pos >= snd_pcm_lib_buffer_bytes(substream)) + new_pos = 0; + WRITE_ONCE(prtd->pos, new_pos); snd_pcm_period_elapsed(substream); } @@ -235,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); + return bytes_to_frames(substream->runtime, READ_ONCE(prtd->pos)); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); -- 2.7.4

2 1

[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 10 Sep '18

10 Sep '18

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..29f577e6dfc0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

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Alsa-devel August 2018