March 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

2 2

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 17 Jun '25

17 Jun '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

2 1

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

5 18

[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

4 4

[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

3 2

[alsa-devel] [PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 24 Nov '22

24 Nov '22

(This series depends on https://lore.kernel.org/patchwork/patch/1126819/ which has not applied to the maintainer's tree yet.) This series makes mt8183-mt6358-ts3a227-max98357 support DP (display port) audio output. ASoC: hdmi-codec: add PCM trigger operator - add a PCM trigger callback if codec driver is interested in PCM events. drm: bridge: it6505: bridge to hdmi-codec - add audio feature support of it6505. - bridge to hdmi-codec so that ASoC can use the audio feature. - defer the "enabling" after receiving PCM START or RESUME event. ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec - bind to hdmi-codec in mt8183-mt6358-ts3a227-max98357. drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting - use the mechanism from the following patch https://mailman.alsa-project.org/pipermail/alsa-devel/2019-September/155302… to report jack status (from DRM to ASoC). Allen Chen (1): drm: bridge: it6505: bridge to hdmi-codec Tzung-Bi Shih (5): ASoC: hdmi-codec: add PCM trigger operator ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 + drivers/gpu/drm/bridge/ite-it6505.c | 191 +++++++++++++++++- include/sound/hdmi-codec.h | 13 ++ sound/soc/codecs/hdmi-codec.c | 31 +++ sound/soc/mediatek/Kconfig | 3 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 36 +++- 6 files changed, 266 insertions(+), 10 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 9

[alsa-devel] [PATCH 00/46] ARM: pxa: towards multiplatform support
by Arnd Bergmann 19 Apr '22

19 Apr '22

Hi PXA maintainers, I'm in the process of getting the old ARM platforms to all build in a single kernel. The largest part of that work is changing all the device drivers to no longer require mach/*.h header files. This series does it for arch/pxa/. As with the omap1 and s3c24xx series I sent before, I don't expect this all to be correct in the first version, though a lot of the patches are fairly simple and I did exhaustive compile-time testing on them. Please test if you have the hardware, or review! Arnd Bcc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Bcc: alsa-devel(a)alsa-project.org Bcc: Bartlomiej Zolnierkiewicz <b.zolnierkie(a)samsung.com> Bcc: Brian Norris <computersforpeace(a)gmail.com> Bcc: Daniel Thompson <daniel.thompson(a)linaro.org> Bcc: David Woodhouse <dwmw2(a)infradead.org> Bcc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Bcc: Dominik Brodowski <linux(a)dominikbrodowski.net> Bcc: dri-devel(a)lists.freedesktop.org Bcc: Felipe Balbi <balbi(a)kernel.org> Bcc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Bcc: Guenter Roeck <linux(a)roeck-us.net> Bcc: Jacek Anaszewski <jacek.anaszewski(a)gmail.com> Bcc: Jens Axboe <axboe(a)kernel.dk> Bcc: Jingoo Han <jingoohan1(a)gmail.com> Bcc: Jonathan Cameron <jic23(a)cam.ac.uk> Bcc: Lee Jones <lee.jones(a)linaro.org> Bcc: Lubomir Rintel <lkundrak(a)v3.sk> Bcc: Marek Vasut <marek.vasut(a)gmail.com> Bcc: Mark Brown <broonie(a)kernel.org> Bcc: Michael Turquette <mturquette(a)baylibre.com> Bcc: Miquel Raynal <miquel.raynal(a)bootlin.com> Bcc: Paul Parsons <lost.distance(a)yahoo.com> Bcc: Pavel Machek <pavel(a)ucw.cz> Bcc: Philipp Zabel <philipp.zabel(a)gmail.com> Bcc: "Rafael J. Wysocki" <rjw(a)rjwysocki.net> Bcc: Richard Weinberger <richard(a)nod.at> Bcc: Russell King <linux(a)armlinux.org.uk> Bcc: Sebastian Reichel <sre(a)kernel.org> Bcc: Sergey Lapin <slapin(a)ossfans.org> Bcc: Stephen Boyd <sboyd(a)kernel.org> Bcc: Tomas Cech <sleep_walker(a)suse.com> Bcc: Ulf Hansson <ulf.hansson(a)linaro.org> Bcc: Vignesh Raghavendra <vigneshr(a)ti.com> Bcc: Viresh Kumar <viresh.kumar(a)linaro.org> Bcc: Wim Van Sebroeck <wim(a)linux-watchdog.org> Cc: linux-clk(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-pm(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-watchdog(a)vger.kernel.org Arnd Bergmann (46): ARM: pxa: split mach/generic.h ARM: pxa: make mainstone.h private ARM: pxa: make mach/regs-uart.h private ARM: pxa: remove mach/dma.h ARM: pxa: split up mach/hardware.h ARM: pxa: stop using mach/bitfield.h ARM: pxa: move mach/sound.h to linux/platform_data/ ARM: pxa: move regs-lcd.h into driver watchdog: sa1100: use platform device registration ARM: pxa: pxa2xx-ac97-lib: use IRQ resource ARM: pxa: cmx270: use platform device for nand ARM: pxa: make addr-map.h header local ARM: pxa: move pcmcia board data into mach-pxa ARM: pxa: use pdev resource for palmld mmio ARM: pxa: maybe fix gpio lookup tables ARM: pxa: tosa: use gpio descriptor for audio ARM: pxa: poodle: use platform data for poodle asoc driver ARM: pxa: corgi: use gpio descriptors for audio ARM: pxa: hx4700: use gpio descriptors for audio ARM: pxa: lubbock: pass udc irqs as resource ARM: pxa: spitz: use gpio descriptors for audio ARM: pxa: eseries: use gpio lookup for audio ARM: pxa: z2: use gpio lookup for audio device ARM: pxa: magician: use platform driver for audio ARM: pxa: mainstone-wm97xx: use gpio lookup table ARM: pxa: zylonite: use gpio lookup instead mfp header input: touchscreen: mainstone: fix pxa2xx+pxa3xx configuration input: touchscreen: mainstone: sync with zylonite driver Input: touchscreen: use wrapper for pxa2xx ac97 registers SoC: pxa: use pdev resource for FIFO regs ASoC: pxa: ac97: use normal MMIO accessors ASoC: pxa: i2s: use normal MMIO accessors ARM: pxa: pcmcia: move smemc configuration back to arch ARM: pxa: remove get_clk_frequency_khz() cpufreq: pxa3: move clk register access to clk driver ARM: pxa: move smemc register access from clk to platform ARM: pxa: move clk register definitions to driver video: backlight: tosa: use gpio lookup table power: tosa: simplify probe function ARM: pxa: tosa: use gpio lookup for battery ARM: pxa: move it8152 PCI support into machine ARM: pxa: remove unused mach/bitfield.h ARM: pxa: pci-it8152: add platform checks ARM: mmp: remove tavorevb board support ARM: mmp: rename pxa_register_device ARM: pxa: move plat-pxa to drivers/soc/ arch/arm/Kconfig | 9 - arch/arm/Makefile | 1 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 1 - arch/arm/common/sa1111.c | 5 +- arch/arm/include/asm/hardware/sa1111.h | 2 - arch/arm/mach-mmp/Kconfig | 10 +- arch/arm/mach-mmp/Makefile | 1 - arch/arm/mach-mmp/devices.c | 2 +- arch/arm/mach-mmp/devices.h | 10 +- arch/arm/mach-mmp/mfp.h | 2 +- arch/arm/mach-mmp/mmp2.h | 48 ++--- arch/arm/mach-mmp/pxa168.h | 60 +++--- arch/arm/mach-mmp/pxa910.h | 38 ++-- arch/arm/mach-mmp/tavorevb.c | 113 ----------- arch/arm/mach-mmp/ttc_dkb.c | 6 +- arch/arm/mach-pxa/Kconfig | 8 +- arch/arm/mach-pxa/Makefile | 22 +- .../mach-pxa/{include/mach => }/addr-map.h | 0 .../arm/mach-pxa/balloon3-pcmcia.c | 4 +- arch/arm/mach-pxa/balloon3.c | 4 +- .../mach-pxa/{include/mach => }/balloon3.h | 2 +- arch/arm/mach-pxa/cm-x270.c | 25 +++ arch/arm/mach-pxa/cm-x2xx-pci.c | 2 +- arch/arm/mach-pxa/cm-x2xx.c | 7 +- arch/arm/mach-pxa/cm-x300.c | 12 +- .../arm/mach-pxa/cm_x255-pcmcia.c | 2 +- .../arm/mach-pxa/cm_x270-pcmcia.c | 2 +- .../arm/mach-pxa/cm_x2xx-pcmcia.c | 2 +- arch/arm/mach-pxa/colibri-evalboard.c | 1 - .../arm/mach-pxa/colibri-pcmcia.c | 2 +- arch/arm/mach-pxa/colibri-pxa270-income.c | 1 - arch/arm/mach-pxa/colibri-pxa270.c | 2 +- arch/arm/mach-pxa/colibri-pxa300.c | 3 +- arch/arm/mach-pxa/colibri-pxa320.c | 2 +- arch/arm/mach-pxa/colibri-pxa3xx.c | 1 - arch/arm/mach-pxa/corgi.c | 23 ++- arch/arm/mach-pxa/{include/mach => }/corgi.h | 2 +- arch/arm/mach-pxa/corgi_pm.c | 3 +- arch/arm/mach-pxa/csb726.c | 3 +- arch/arm/mach-pxa/devices.c | 15 +- .../arm/mach-pxa/e740-pcmcia.c | 2 +- arch/arm/mach-pxa/em-x270.c | 2 +- arch/arm/mach-pxa/eseries.c | 34 +++- arch/arm/mach-pxa/ezx.c | 1 - arch/arm/mach-pxa/generic.c | 78 ++++++-- arch/arm/mach-pxa/generic.h | 9 - arch/arm/mach-pxa/gumstix.c | 1 - .../arm/mach-pxa/hx4700-pcmcia.c | 4 +- arch/arm/mach-pxa/hx4700.c | 18 +- arch/arm/mach-pxa/{include/mach => }/hx4700.h | 2 +- arch/arm/mach-pxa/idp.c | 2 - arch/arm/mach-pxa/include/mach/bitfield.h | 114 ----------- arch/arm/mach-pxa/include/mach/dma.h | 17 -- arch/arm/mach-pxa/include/mach/generic.h | 1 - arch/arm/mach-pxa/include/mach/mfp.h | 2 +- arch/arm/mach-pxa/include/mach/pxa-regs.h | 52 +++++ arch/arm/mach-pxa/include/mach/pxa2xx-regs.h | 47 +---- arch/arm/mach-pxa/include/mach/pxa3xx-regs.h | 71 +------ arch/arm/mach-pxa/include/mach/regs-ost.h | 4 +- arch/arm/mach-pxa/include/mach/reset.h | 2 +- arch/arm/mach-pxa/include/mach/tosa.h | 15 -- arch/arm/mach-pxa/irq.c | 3 +- arch/arm/mach-pxa/littleton.c | 1 - arch/arm/mach-pxa/lpd270.c | 4 +- arch/arm/mach-pxa/lubbock.c | 15 +- .../arm/mach-pxa/{include/mach => }/lubbock.h | 2 - arch/arm/mach-pxa/magician.c | 54 ++++- arch/arm/mach-pxa/mainstone.c | 15 +- .../mach-pxa/{include/mach => }/mainstone.h | 2 - arch/arm/mach-pxa/mfp-pxa2xx.c | 1 + arch/arm/mach-pxa/mfp-pxa2xx.h | 2 +- arch/arm/mach-pxa/mfp-pxa3xx.c | 1 - arch/arm/mach-pxa/mfp-pxa3xx.h | 2 +- arch/arm/mach-pxa/mioa701.c | 2 +- arch/arm/mach-pxa/mxm8x10.c | 8 +- arch/arm/mach-pxa/palm27x.c | 2 +- .../arm/mach-pxa/palmld-pcmcia.c | 5 +- arch/arm/mach-pxa/palmld.c | 23 ++- arch/arm/mach-pxa/{include/mach => }/palmld.h | 2 +- arch/arm/mach-pxa/palmt5.c | 11 +- .../arm/mach-pxa/palmtc-pcmcia.c | 4 +- arch/arm/mach-pxa/palmtc.c | 4 +- arch/arm/mach-pxa/{include/mach => }/palmtc.h | 2 +- arch/arm/mach-pxa/palmte2.c | 2 +- arch/arm/mach-pxa/palmtreo.c | 2 +- .../arm/mach-pxa/palmtx-pcmcia.c | 4 +- arch/arm/mach-pxa/palmtx.c | 13 +- arch/arm/mach-pxa/{include/mach => }/palmtx.h | 2 +- arch/arm/mach-pxa/palmz72.c | 2 +- .../it8152.c => mach-pxa/pci-it8152.c} | 25 ++- .../it8152.h => mach-pxa/pci-it8152.h} | 2 - arch/arm/mach-pxa/pcm990-baseboard.c | 2 +- arch/arm/mach-pxa/poodle.c | 31 ++- arch/arm/mach-pxa/{include/mach => }/poodle.h | 4 +- arch/arm/mach-pxa/pxa-regs.h | 1 + arch/arm/mach-pxa/pxa25x.c | 6 +- arch/arm/mach-pxa/pxa25x.h | 2 +- arch/arm/mach-pxa/pxa27x-udc.h | 2 + arch/arm/mach-pxa/pxa27x.c | 6 +- arch/arm/mach-pxa/pxa27x.h | 2 +- arch/arm/mach-pxa/pxa2xx.c | 1 - arch/arm/mach-pxa/pxa300.c | 1 + arch/arm/mach-pxa/pxa320.c | 1 + arch/arm/mach-pxa/pxa3xx-ulpi.c | 2 +- arch/arm/mach-pxa/pxa3xx.c | 11 +- arch/arm/mach-pxa/pxa3xx.h | 2 +- arch/arm/mach-pxa/pxa930.c | 1 + arch/arm/mach-pxa/regs-rtc.h | 2 +- arch/arm/mach-pxa/regs-u2d.h | 2 - .../mach-pxa/{include/mach => }/regs-uart.h | 2 + arch/arm/mach-pxa/reset.c | 3 - arch/arm/mach-pxa/sleep.S | 5 +- arch/arm/mach-pxa/smemc.c | 11 +- arch/arm/mach-pxa/spitz.c | 33 ++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 2 +- arch/arm/mach-pxa/spitz_pm.c | 3 +- arch/arm/mach-pxa/standby.S | 1 - .../arm/mach-pxa/stargate2-pcmcia.c | 2 +- arch/arm/mach-pxa/tosa.c | 63 +++++- .../arm/mach-pxa/trizeps4-pcmcia.c | 4 +- arch/arm/mach-pxa/trizeps4.c | 4 +- .../mach-pxa/{include/mach => }/trizeps4.h | 3 +- .../arm/mach-pxa/viper-pcmcia.c | 6 +- .../arm/mach-pxa/viper-pcmcia.h | 0 arch/arm/mach-pxa/viper.c | 8 +- .../arm/mach-pxa/vpac270-pcmcia.c | 4 +- arch/arm/mach-pxa/vpac270.c | 4 +- .../arm/mach-pxa/{include/mach => }/vpac270.h | 0 arch/arm/mach-pxa/xcep.c | 2 +- arch/arm/mach-pxa/z2.c | 11 + arch/arm/mach-pxa/zeus.c | 6 +- arch/arm/mach-pxa/zylonite.c | 34 +++- arch/arm/mach-pxa/zylonite.h | 2 + arch/arm/mach-pxa/zylonite_pxa300.c | 1 + arch/arm/mach-pxa/zylonite_pxa320.c | 1 + arch/arm/mach-sa1100/generic.c | 6 +- arch/arm/mach-sa1100/include/mach/reset.h | 1 - drivers/ata/pata_palmld.c | 3 +- drivers/clk/pxa/clk-pxa.c | 8 +- drivers/clk/pxa/clk-pxa.h | 9 +- drivers/clk/pxa/clk-pxa25x.c | 46 ++--- drivers/clk/pxa/clk-pxa27x.c | 68 +++---- drivers/clk/pxa/clk-pxa3xx.c | 139 +++++++++++-- drivers/cpufreq/pxa2xx-cpufreq.c | 6 +- drivers/cpufreq/pxa3xx-cpufreq.c | 65 +++--- drivers/input/mouse/pxa930_trkball.c | 1 - drivers/input/touchscreen/Kconfig | 2 + drivers/input/touchscreen/mainstone-wm97xx.c | 123 ++++++------ drivers/input/touchscreen/zylonite-wm97xx.c | 34 ++-- drivers/leds/leds-locomo.c | 1 - drivers/mmc/host/pxamci.c | 2 +- drivers/mtd/maps/pxa2xx-flash.c | 2 - drivers/mtd/nand/raw/cmx270_nand.c | 89 +++------ drivers/pcmcia/Makefile | 16 -- drivers/pcmcia/pxa2xx_base.c | 48 ++--- drivers/pcmcia/pxa2xx_sharpsl.c | 3 +- drivers/pcmcia/sa1111_generic.c | 1 - drivers/pcmcia/sa1111_lubbock.c | 1 - drivers/pcmcia/soc_common.c | 2 - drivers/pcmcia/soc_common.h | 120 +---------- drivers/power/supply/tosa_battery.c | 189 ++++++++++-------- drivers/rtc/rtc-pxa.c | 2 - drivers/soc/Kconfig | 1 + drivers/soc/Makefile | 1 + .../arm/plat-pxa => drivers/soc/pxa}/Kconfig | 5 +- .../arm/plat-pxa => drivers/soc/pxa}/Makefile | 4 - {arch/arm/plat-pxa => drivers/soc/pxa}/mfp.c | 2 +- {arch/arm/plat-pxa => drivers/soc/pxa}/ssp.c | 0 drivers/usb/gadget/udc/pxa25x_udc.c | 37 ++-- drivers/usb/gadget/udc/pxa25x_udc.h | 7 +- drivers/usb/host/ohci-pxa27x.c | 3 +- drivers/video/backlight/tosa_bl.c | 10 +- drivers/video/backlight/tosa_bl.h | 8 + drivers/video/backlight/tosa_lcd.c | 28 ++- .../video/fbdev/pxa3xx-regs.h | 24 +-- drivers/video/fbdev/pxafb.c | 4 +- drivers/watchdog/sa1100_wdt.c | 88 +++++--- include/linux/clk/pxa.h | 16 ++ include/linux/platform_data/asoc-poodle.h | 16 ++ .../linux/platform_data/asoc-pxa.h | 4 +- include/linux/platform_data/video-pxafb.h | 22 +- .../hardware.h => include/linux/soc/pxa/cpu.h | 61 +----- .../plat => include/linux/soc/pxa}/mfp.h | 6 +- include/linux/soc/pxa/smemc.h | 13 ++ include/pcmcia/soc_common.h | 125 ++++++++++++ include/sound/pxa2xx-lib.h | 4 + sound/arm/pxa2xx-ac97-lib.c | 145 +++++++++----- .../arm/pxa2xx-ac97-regs.h | 42 ++-- sound/arm/pxa2xx-ac97.c | 3 +- sound/soc/pxa/corgi.c | 42 ++-- sound/soc/pxa/e740_wm9705.c | 37 ++-- sound/soc/pxa/e750_wm9705.c | 33 ++- sound/soc/pxa/e800_wm9712.c | 33 ++- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/hx4700.c | 34 ++-- sound/soc/pxa/magician.c | 141 ++++--------- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 51 ++--- sound/soc/pxa/pxa2xx-ac97.c | 24 ++- sound/soc/pxa/pxa2xx-i2s.c | 112 ++++++----- sound/soc/pxa/spitz.c | 58 +++--- sound/soc/pxa/tosa.c | 18 +- sound/soc/pxa/z2.c | 8 +- 205 files changed, 1916 insertions(+), 1795 deletions(-) delete mode 100644 arch/arm/mach-mmp/tavorevb.c rename arch/arm/mach-pxa/{include/mach => }/addr-map.h (100%) rename drivers/pcmcia/pxa2xx_balloon3.c => arch/arm/mach-pxa/balloon3-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/balloon3.h (99%) rename drivers/pcmcia/pxa2xx_cm_x255.c => arch/arm/mach-pxa/cm_x255-pcmcia.c (98%) rename drivers/pcmcia/pxa2xx_cm_x270.c => arch/arm/mach-pxa/cm_x270-pcmcia.c (98%) rename drivers/pcmcia/pxa2xx_cm_x2xx.c => arch/arm/mach-pxa/cm_x2xx-pcmcia.c (96%) rename drivers/pcmcia/pxa2xx_colibri.c => arch/arm/mach-pxa/colibri-pcmcia.c (99%) rename arch/arm/mach-pxa/{include/mach => }/corgi.h (98%) rename drivers/pcmcia/pxa2xx_e740.c => arch/arm/mach-pxa/e740-pcmcia.c (98%) rename drivers/pcmcia/pxa2xx_hx4700.c => arch/arm/mach-pxa/hx4700-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/hx4700.h (99%) delete mode 100644 arch/arm/mach-pxa/include/mach/bitfield.h delete mode 100644 arch/arm/mach-pxa/include/mach/dma.h delete mode 100644 arch/arm/mach-pxa/include/mach/generic.h create mode 100644 arch/arm/mach-pxa/include/mach/pxa-regs.h rename arch/arm/mach-pxa/{include/mach => }/lubbock.h (97%) rename arch/arm/mach-pxa/{include/mach => }/mainstone.h (99%) rename drivers/pcmcia/pxa2xx_palmld.c => arch/arm/mach-pxa/palmld-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmld.h (98%) rename drivers/pcmcia/pxa2xx_palmtc.c => arch/arm/mach-pxa/palmtc-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmtc.h (98%) rename drivers/pcmcia/pxa2xx_palmtx.c => arch/arm/mach-pxa/palmtx-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/palmtx.h (98%) rename arch/arm/{common/it8152.c => mach-pxa/pci-it8152.c} (95%) rename arch/arm/{include/asm/hardware/it8152.h => mach-pxa/pci-it8152.h} (98%) rename arch/arm/mach-pxa/{include/mach => }/poodle.h (97%) create mode 100644 arch/arm/mach-pxa/pxa-regs.h rename arch/arm/mach-pxa/{include/mach => }/regs-uart.h (99%) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (99%) rename drivers/pcmcia/pxa2xx_stargate2.c => arch/arm/mach-pxa/stargate2-pcmcia.c (99%) rename drivers/pcmcia/pxa2xx_trizeps4.c => arch/arm/mach-pxa/trizeps4-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/trizeps4.h (98%) rename drivers/pcmcia/pxa2xx_viper.c => arch/arm/mach-pxa/viper-pcmcia.c (97%) rename include/linux/platform_data/pcmcia-pxa2xx_viper.h => arch/arm/mach-pxa/viper-pcmcia.h (100%) rename drivers/pcmcia/pxa2xx_vpac270.c => arch/arm/mach-pxa/vpac270-pcmcia.c (98%) rename arch/arm/mach-pxa/{include/mach => }/vpac270.h (100%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/Kconfig (83%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/Makefile (51%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/mfp.c (99%) rename {arch/arm/plat-pxa => drivers/soc/pxa}/ssp.c (100%) create mode 100644 drivers/video/backlight/tosa_bl.h rename arch/arm/mach-pxa/include/mach/regs-lcd.h => drivers/video/fbdev/pxa3xx-regs.h (90%) create mode 100644 include/linux/clk/pxa.h create mode 100644 include/linux/platform_data/asoc-poodle.h rename arch/arm/mach-pxa/include/mach/audio.h => include/linux/platform_data/asoc-pxa.h (93%) rename arch/arm/mach-pxa/include/mach/hardware.h => include/linux/soc/pxa/cpu.h (75%) rename {arch/arm/plat-pxa/include/plat => include/linux/soc/pxa}/mfp.h (98%) create mode 100644 include/linux/soc/pxa/smemc.h create mode 100644 include/pcmcia/soc_common.h rename arch/arm/mach-pxa/include/mach/regs-ac97.h => sound/arm/pxa2xx-ac97-regs.h (71%) -- 2.20.0

9 53

[alsa-devel] [PATCH v7 0/8] Fix year 2038 issue for sound subsystem
by Arnd Bergmann 19 Oct '21

19 Oct '21

This is a series I worked on with Baolin in 2017 and 2018, but we never quite managed to finish up the last pieces. During the ALSA developer meetup at ELC-E 2018 in Edinburgh, a decision was made to go with this approach for keeping best compatibility with existing source code, and then I failed to follow up by resending the patches. Now I have patches for all remaining time_t uses in the kernel, so it's absolutely time to revisit them. I have done more review of the patches myself and found a couple of minor issues that I have fixed up, otherwise the series is still the same as before. Conceptually, the idea of these patches is: - 64-bit applications should see no changes at all, neither compile-time nor run-time. - 32-bit code compiled with a 64-bit time_t currently does not work with ALSA, and requires kernel changes and/or sound/asound.h changes - Most 32-bit code using these interfaces will work correctly on a modified kernel, with or without the uapi header changes. - 32-bit code using SNDRV_TIMER_IOCTL_TREAD requires the updated header file for 64-bit time_t support - 32-bit i386 user space with 64-bit time_t is broken for SNDRV_PCM_IOCTL_STATUS, SNDRV_RAWMIDI_IOCTL_STATUS and SNDRV_PCM_IOCTL_SYNC_PTR because of i386 alignment. This is also addressed by the updated uapi header. - PCM mmap is currently supported on native x86 kernels (both 32-bit and 64-bit) but not for compat mode. This series breaks the 32-bit native mmap support for 32-bit time_t, but instead allows it for 64-bit time_t on both native and compat kernels. This seems to be the best trade-off, as mmap support is optional already, and most 32-bit code runs in compat mode anyway. - I've tried to avoid breaking compilation of 32-bit code as much as possible. Anything that does break however is likely code that is already broken on 64-bit time_t and needs source changes to fix them. I hope I addressed all review comments by now, so please pull this for linux-5.6. A git branch with the same contents is available for testing at [1]. Arnd [1] https://git.kernel.org/pub/scm/linux/kernel/git/arnd/playground.git y2038-alsa-v7 [2] https://lore.kernel.org/lkml/CAK8P3a2Os66+iwQYf97qh05W2JP8rmWao8zmKoHiXqVHv… Changes since v6: (Arnd): - Add a patch to update the API versions - Hide a timespec reference in #ifndef __KERNEL__ to remove the last reference to time_t - Use a more readable way to do padding and describe it in the changelog - Rebase to linux-5.5-rc1, changing include/sound/soc-component.h and sound/drivers/aloop.c as needed. Changes since v5 (Arnd): - Rebased to linux-5.4-rc4 - Updated to completely remove timespec and time_t references from alsa - found and fixed a few bugs Changes since v4 (Baolin): - Add patch 5 to change trigger_tstamp member of struct snd_pcm_runtime. - Add patch 8 to change internal timespec. - Add more explanation in commit message. - Use ktime_get_real_ts64() in patch 6. - Split common code out into a separate function in patch 6. - Fix tu->tread bug in patch 6 and remove #if __BITS_PER_LONG == 64 macro. Changes since v3: - Move struct snd_pcm_status32 to pcm.h file. - Modify comments and commit message. - Add new patch2 ~ patch6. Changes since v2: - Renamed all structures to make clear. - Remove CONFIG_X86_X32 macro and introduced new compat_snd_pcm_status64_x86_32. Changes since v1: - Add one macro for struct snd_pcm_status_32 which only active in 32bits kernel. - Convert pcm_compat.c to use struct snd_pcm_status_64. - Convert pcm_native.c to use struct snd_pcm_status_64. --- Arnd Bergmann (3): ALSA: move snd_pcm_ioctl_sync_ptr_compat into pcm_native.c ALSA: add new 32-bit layout for snd_pcm_mmap_status/control ALSA: bump uapi version numbers Baolin Wang (6): ALSA: Replace timespec with timespec64 ALSA: Avoid using timespec for struct snd_timer_status ALSA: Avoid using timespec for struct snd_ctl_elem_value ALSA: Avoid using timespec for struct snd_pcm_status ALSA: Avoid using timespec for struct snd_rawmidi_status ALSA: Avoid using timespec for struct snd_timer_tread include/sound/pcm.h | 74 ++++++-- include/sound/soc-component.h | 4 +- include/sound/timer.h | 4 +- include/uapi/sound/asound.h | 145 +++++++++++++-- sound/core/pcm.c | 12 +- sound/core/pcm_compat.c | 282 ++++++++---------------------- sound/core/pcm_lib.c | 38 ++-- sound/core/pcm_native.c | 226 +++++++++++++++++++++--- sound/core/rawmidi.c | 132 +++++++++++--- sound/core/rawmidi_compat.c | 87 +++------ sound/core/timer.c | 229 ++++++++++++++++++------ sound/core/timer_compat.c | 62 +------ sound/drivers/aloop.c | 2 +- sound/pci/hda/hda_controller.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 4 +- 15 files changed, 817 insertions(+), 494 deletions(-) -- 2.20.0

5 46

ASoc: soc_core.c stream direction from snd_soc_dai
by Matt Flax 26 Feb '21

26 Feb '21

Hi there, A large number of audio codecs allow different formats for playback and capture. This becomes very useful when there are different latencies between playback and capture hardware data lines. For example digital isolation chips typically have a 1 bit delay in propagation as the bit clock rate gets faster for higher sample rates. By setting the capture and playback formats to differ by one or two bit clock cycles, the delay problem is solved. There doesn't seem to be a simple way to detect stream direction in the codec driver's set_fmt function. The snd_soc_runtime_set_dai_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-core.c#L1480 calls the snd_soc_dai_set_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-dai.c#L101 which calls the set_fmt function : https://github.com/torvalds/linux/blob/master/include/sound/soc-dai.h#L189 The snd_soc_dai_ops set_fmt function is defined as : int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); Is there a simple way to find the stream direction from a snd_soc_dai ? If the stream direction can be detected then the playback and capture formats can be set independently for the codec. It there a different way to set the playback and capture formats for the codec independently at runtime, depending on the sample rate ? Matt

5 8

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Disable quirks for BOSS Katana amplifiers
by Takashi Iwai 16 Oct '20

16 Oct '20

On Fri, 11 Oct 2019 19:19:36 +0200, Szabolcs Szőke wrote: > > BOSS Katana amplifiers cannot be used for recording or playback if quirks > are applied > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195223 > Signed-off-by: Szabolcs Szőke <szszoke.code(a)gmail.com> Applied now. Thanks. Takashi > > --- > sound/usb/pcm.c | 3 +++ > 1 file changed, 3 insertions(+) > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index 33cd26763c0e..daadb0c66eee 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -348,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, > ep = 0x84; > ifnum = 0; > goto add_sync_ep_from_ifnum; > + case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ > + /* BOSS Katana amplifiers do not need quirks */ > + return 0; > } > > if (attr == USB_ENDPOINT_SYNC_ASYNC && > -- > 2.20.1 >

5 20

[alsa-devel] [PATCH] ALSA: usb-audio: fix spelling mistake "Frequence" -> "Frequency"
by Colin King 02 Oct '20

02 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in equalizer name fields, fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer_us16x08.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 26ed23b18b77..4f594f9552a8 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -1119,7 +1119,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ Low Frequence", + .name = "EQ Low Frequency", }, { /* EQ mid low gain */ .kcontrol_new = &snd_us16x08_eq_gain_ctl, @@ -1133,7 +1133,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidLow Frequence", + .name = "EQ MidLow Frequency", }, { /* EQ mid low Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1154,7 +1154,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidHigh Frequence", + .name = "EQ MidHigh Frequency", }, { /* EQ mid high Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1175,7 +1175,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ High Frequence", + .name = "EQ High Frequency", }, }; -- 2.19.1

3 3

[alsa-devel] Attempting to understand odd snd-usb-audio code and behavior
by Keith A. Milner 21 Aug '20

21 Aug '20

Over the last few weeks I've been trying to dig through the snd-usb-audio code to try to get an understanding of it. I've used ftrace and also a local version which I have heavily polluted with kprintf(KERN_INFO statements so I can track what it's doing in practice. I've been playing with this with some USB audio interfaces I have some which work, and some which don't, and trying to understand the differences and, ultimately, what I can do to make the non-functioning ones work. I've got to the point where I have a fairly good understanding of some parts, but I'm struggling to understand why the code is behaving as it does in practice. My observation when plugging the USB device in is that, as expected, it tries to build the audio device, and add components to it including attaching capture and playback streams. It seems to go through this phase without too many surprises, and we get to the standard dmesg output: usbcore: registered new interface driver snd-usb-audio After this, it gets confusing. It seems to open and close the PCMs (e.g, snd_usb_capture_open) several times in a row. Each time it sets up the hardware info, checks and sets the format and then closes the PCM. I don't see anything obviously different between most of these cycles. Here's an example my logging output (hopefully this will make sense): Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Setting up hw info Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Checking format: Aug 22 20:04:16 kerneldev kernel: [ 923.140790] Interface number = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting index = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] format type = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] frame size = 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140791] Setting pt=125 ptmin=125 Aug 22 20:04:16 kerneldev kernel: [ 923.140792] Checking for unusual bit rates Aug 22 20:04:16 kerneldev kernel: [ 923.140794] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.140921] Called: pcm.c - snd_usb_hw_params for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Setting format: Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Bytes per period = 8816 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Frames per period = 1102 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Periods per buffer = 4 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Channels = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Rate = 44100 Aug 22 20:04:16 kerneldev kernel: [ 923.141279] Called: endpoint.c - snd_usb_add_endpoint of type 0 with direction 1 Aug 22 20:04:16 kerneldev kernel: [ 923.141292] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141293] Called: pcm.c - configure_endpoint Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Setting endpoint params for endpoint num 142 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] channels=2 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] period bytes=8816 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] frames per period=1102 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] periods per buffer=4 Aug 22 20:04:16 kerneldev kernel: [ 923.141296] Called: endpoint.c - data_ep_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141301] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142768] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142772] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142774] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142830] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142836] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142837] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142846] Called: pcm.c - snd_usb_capture_close for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146760] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146834] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146838] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146841] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146845] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146848] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Setting up hw info It seems to do this several times, before the behavior changing slightly, as it now starts to try to set the endpoint parameters and seems to leave the stream open. However, it then tries to open the stream and set the endpoint parameters again resulting in errors: Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Setting endpoint params for endpoint num 142 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] channels=2 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] period bytes=1768 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] frames per period=221 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] periods per buffer=593 Aug 22 20:22:00 kerneldev kernel: [ 1987.452947] usb 1-1: Unable to change format on ep #8e: already in use Maybe this is driven by hotplug or something, but it seems to be performing a lot of pointless activities. Eventually it seems to try to open the PCM playback or capture stream (with some devices it opens playback, with others capture) and then checks the pointer: Aug 22 20:22:00 kerneldev kernel: [ 1987.470194] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470656] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470657] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Pointer is 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473567] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473569] Pointer is 0 The above is from a non-working device. For a working device the pointer increments. Eventually this activity ends: Aug 22 20:22:05 kerneldev kernel: [ 1992.473575] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473576] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473578] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473602] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503287] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503288] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503323] Called: pcm.c - snd_usb_playback_close for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503656] Called: init.c - snd_card_file_remove This seems as if it might be some sort of test of the PCMs, to make sure they are working. Certainly in non-working devices the pcm_pointer stays at zero and in working devices it increments. It also seems to try harder with some devices than others, even though the device descriptors seem very similar. Can anyone explain any of this to me, particularly why it's exhibiting this behaviour. I've uploaded one of the logs from a non-working device to Pastebin if anyone wants to look at a full example: https://pastebin.com/uBZEuQ57 Also, is this the best way to approach this, or are there other tools/techniques that can help? I've experimented with ftrace and that was useful, but didn't give the level of detail I wanted. Cheers, Keith

3 10

[PATCH] ASoC: Intel: haswell: Power transition refactor
by Cezary Rojewski 20 Jun '20

20 Jun '20

Update D0 <-> D3 sequence to correctly transition hardware and DSP core from and to D3. On top of that, set SHIM registers to their recommended defaults during D0 and D3 proceduces as HW does not reset registers for us. Connected to: [alsa-devel][BUG] bdw-rt5650 DSP boot timeout https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153098.html Github issue ticket reference: https://github.com/thesofproject/linux/pull/1842 Tested on: - BDW-Y RVP with rt286 - SAMUS with rt5677 Proposed solution (both in July 2019 and on github): 'Revert "ASoC: Intel: Work around to fix HW d3 potential crash issue"' is NAKed as it only covers the problem up and actually brings back the undefined behavior: some registers (e.g.: APLLSE) are describing LPT offsets rather than WPT ones. In consequence, during power-transitions driver issues incorrect writes and leaves the regs of interest alone. Existing patch - the non-revert - does not resolve the HW D3 issue at all as it ignores the recommended sequence and does not initialize hardware registers as expected. And thus, leaving things as are is also unacceptable. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++++---------- 1 file changed, 104 insertions(+), 81 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index 88c3f63bded9..de80e19454c1 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -243,45 +243,92 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } +#define CSR_DEFAULT_VALUE 0x8480040E +#define ISC_DEFAULT_VALUE 0x0 +#define ISD_DEFAULT_VALUE 0x0 +#define IMC_DEFAULT_VALUE 0x7FFF0003 +#define IMD_DEFAULT_VALUE 0x7FFF0003 +#define IPCC_DEFAULT_VALUE 0x0 +#define IPCD_DEFAULT_VALUE 0x0 +#define CLKCTL_DEFAULT_VALUE 0x7FF +#define CSR2_DEFAULT_VALUE 0x0 +#define LTR_CTRL_DEFAULT_VALUE 0x0 +#define HMD_CTRL_DEFAULT_VALUE 0x0 + +static void hsw_set_shim_defaults(struct sst_dsp *sst) +{ + sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE); +} + +/* all clock-gating minus DCLCGE and DTCGE */ +#define SST_VDRTCL2_CG_OTHER 0xB7D + static void hsw_set_dsp_D3(struct sst_dsp *sst) { - u32 val; u32 reg; - /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + /* disable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + reg &= ~(SST_VDRTCL2_DCLCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* enable power gating and switch off DRAM & IRAM blocks */ - val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - val |= SST_VDRTCL0_DSRAMPGE_MASK | - SST_VDRTCL0_ISRAMPGE_MASK; - val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); - writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); + /* stall, reset and set 24MHz XOSC */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST); - /* switch off audio PLL */ - val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - val |= SST_VDRTCL2_APLLSE_MASK; - writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + /* DRAM power gating all */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_ISRAMPGE_MASK | + SST_VDRTCL0_DSRAMPGE_MASK; + reg &= ~(SST_VDRTCL0_D3SRAMPGD); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + udelay(50); - /* disable MCLK(clkctl.smos = 0) */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, 0); + /* PLL shutdown enable */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_APLLSE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Set D3 state, delay 50 us */ - val = readl(sst->addr.pci_cfg + SST_PMCS); - val |= SST_PMCS_PS_MASK; - writel(val, sst->addr.pci_cfg + SST_PMCS); - udelay(50); + /* disable MCLK */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK, 0); - /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + /* switch clock gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_CG_OTHER; + reg &= ~(SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* enable DTCGE separatelly */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + reg |= SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* set shim defaults */ + hsw_set_shim_defaults(sst); + + /* set D3 */ + reg = readl(sst->addr.pci_cfg + SST_PMCS); + reg |= SST_PMCS_PS_MASK; + writel(reg, sst->addr.pci_cfg + SST_PMCS); udelay(50); + /* enable clock core gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + udelay(50); } static void hsw_reset(struct sst_dsp *sst) @@ -299,75 +346,62 @@ static void hsw_reset(struct sst_dsp *sst) SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); } +/* recommended CSR state for power-up */ +#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK) + static int hsw_set_dsp_D0(struct sst_dsp *sst) { - int tries = 10; - u32 reg, fw_dump_bit; + u32 reg; - /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + /* disable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + reg &= ~(SST_VDRTCL2_DCLCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* switch clock gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_CG_OTHER; + reg &= ~(SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Set D0 state */ + /* set D0 */ reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg &= ~SST_PMCS_PS_MASK; + reg &= ~(SST_PMCS_PS_MASK); writel(reg, sst->addr.pci_cfg + SST_PMCS); - /* check that ADSP shim is enabled */ - while (tries--) { - reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; - if (reg == 0) - goto finish; - - msleep(1); - } - - return -ENODEV; - -finish: - /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + /* DRAM power gating none */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK | + SST_VDRTCL0_DSRAMPGE_MASK); + reg |= SST_VDRTCL0_D3SRAMPGD; + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + mdelay(10); - /* stall DSP core, set clk to 192/96Mhz */ - sst_dsp_shim_update_bits_unlocked(sst, - SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, - SST_CSR_STALL | SST_CSR_DCS(4)); + /* set shim defaults */ + hsw_set_shim_defaults(sst); - /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + /* restore MCLK */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, - SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + SST_CLKCTL_MASK, SST_CLKCTL_MASK); - /* Stall and reset core, set CSR */ - hsw_reset(sst); - - /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + /* PLL shutdown disable */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + reg &= ~(SST_VDRTCL2_APLLSE_MASK); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 | + SST_CSR_STALL | SST_CSR_DCS(4)); udelay(50); - /* switch on audio PLL */ + /* enable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~SST_VDRTCL2_APLLSE_MASK; + reg |= SST_VDRTCL2_DCLCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* set default power gating control, enable power gating control for all blocks. that is, - can't be accessed, please enable each block before accessing. */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; - /* for D0, always enable the block(DSRAM[0]) used for FW dump */ - fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; - writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); - + /* clear reset */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0); /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, @@ -384,12 +418,6 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); - /* clear IPC registers */ - sst_dsp_shim_write(sst, SST_IPCX, 0x0); - sst_dsp_shim_write(sst, SST_IPCD, 0x0); - sst_dsp_shim_write(sst, 0x80, 0x6); - sst_dsp_shim_write(sst, 0xe0, 0x300a); - return 0; } @@ -415,11 +443,6 @@ static void hsw_sleep(struct sst_dsp *sst) { dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); - /* put DSP into reset and stall */ - sst_dsp_shim_update_bits(sst, SST_CSR, - SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, - SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); - hsw_set_dsp_D3(sst); dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); } -- 2.17.1

5 11

[alsa-devel] [PATCH 0/2 v4] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 12 Jun '20

12 Jun '20

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +++-- include/uapi/sound/asound.h | 6 ++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 +++--- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 97 ++++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 110 insertions(+), 19 deletions(-) -- 2.13.6

4 17

[PATCH v2 0/2] powerpc: Remove support for ppc405/440 Xilinx platforms
by Michal Simek 03 Jun '20

03 Jun '20

Hi, recently we wanted to update xilinx intc driver and we found that function which we wanted to remove is still wired by ancient Xilinx PowerPC platforms. Here is the thread about it. https://lore.kernel.org/linux-next/48d3232d-0f1d-42ea-3109-f44bbabfa2e8@xil… I have been talking about it internally and there is no interest in these platforms and it is also orphan for quite a long time. None is really running/testing these platforms regularly that's why I think it makes sense to remove them also with drivers which are specific to this platform. U-Boot support was removed in 2017 without anybody complain about it https://github.com/Xilinx/u-boot-xlnx/commit/98f705c9cefdfdba62c069821bbba1… Based on current ppc/next. If anyone has any objection about it, please let me know. Thanks, Michal Changes in v2: - Remove also sound/drivers/pcm-indirect2.[ch] files Reported-by: Takashi Iwai <tiwai(a)suse.de> - Based on my chat with Arnd I removed arch/powerpc/xmon/ changes done in v1 to keep them the same as before. (kbuild reported some issues with it too) Michal Simek (2): sound: ac97: Remove sound driver for ancient platform powerpc: Remove Xilinx PPC405/PPC440 support Documentation/devicetree/bindings/xilinx.txt | 143 -- Documentation/powerpc/bootwrapper.rst | 28 +- MAINTAINERS | 6 - arch/powerpc/Kconfig.debug | 2 +- arch/powerpc/boot/Makefile | 7 +- arch/powerpc/boot/dts/Makefile | 1 - arch/powerpc/boot/dts/virtex440-ml507.dts | 406 ------ arch/powerpc/boot/dts/virtex440-ml510.dts | 466 ------- arch/powerpc/boot/ops.h | 1 - arch/powerpc/boot/serial.c | 5 - arch/powerpc/boot/uartlite.c | 79 -- arch/powerpc/boot/virtex.c | 97 -- arch/powerpc/boot/virtex405-head.S | 31 - arch/powerpc/boot/wrapper | 8 - arch/powerpc/configs/40x/virtex_defconfig | 75 - arch/powerpc/configs/44x/virtex5_defconfig | 74 - arch/powerpc/configs/ppc40x_defconfig | 8 - arch/powerpc/configs/ppc44x_defconfig | 8 - arch/powerpc/include/asm/xilinx_intc.h | 16 - arch/powerpc/include/asm/xilinx_pci.h | 21 - arch/powerpc/kernel/cputable.c | 39 - arch/powerpc/platforms/40x/Kconfig | 31 - arch/powerpc/platforms/40x/Makefile | 1 - arch/powerpc/platforms/40x/virtex.c | 54 - arch/powerpc/platforms/44x/Kconfig | 37 - arch/powerpc/platforms/44x/Makefile | 2 - arch/powerpc/platforms/44x/virtex.c | 60 - arch/powerpc/platforms/44x/virtex_ml510.c | 30 - arch/powerpc/platforms/Kconfig | 4 - arch/powerpc/sysdev/Makefile | 2 - arch/powerpc/sysdev/xilinx_intc.c | 88 -- arch/powerpc/sysdev/xilinx_pci.c | 132 -- drivers/char/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 2 +- sound/drivers/Kconfig | 12 - sound/drivers/Makefile | 2 - sound/drivers/ml403-ac97cr.c | 1298 ------------------ sound/drivers/pcm-indirect2.c | 560 -------- sound/drivers/pcm-indirect2.h | 127 -- 39 files changed, 7 insertions(+), 3958 deletions(-) delete mode 100644 arch/powerpc/boot/dts/virtex440-ml507.dts delete mode 100644 arch/powerpc/boot/dts/virtex440-ml510.dts delete mode 100644 arch/powerpc/boot/uartlite.c delete mode 100644 arch/powerpc/boot/virtex.c delete mode 100644 arch/powerpc/boot/virtex405-head.S delete mode 100644 arch/powerpc/configs/40x/virtex_defconfig delete mode 100644 arch/powerpc/configs/44x/virtex5_defconfig delete mode 100644 arch/powerpc/include/asm/xilinx_intc.h delete mode 100644 arch/powerpc/include/asm/xilinx_pci.h delete mode 100644 arch/powerpc/platforms/40x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex_ml510.c delete mode 100644 arch/powerpc/sysdev/xilinx_intc.c delete mode 100644 arch/powerpc/sysdev/xilinx_pci.c delete mode 100644 sound/drivers/ml403-ac97cr.c delete mode 100644 sound/drivers/pcm-indirect2.c delete mode 100644 sound/drivers/pcm-indirect2.h -- 2.26.0

5 12

[PATCH 0/2] powerpc: Remove support for ppc405/440 Xilinx platforms
by Michal Simek 21 May '20

21 May '20

Hi, recently we wanted to update xilinx intc driver and we found that function which we wanted to remove is still wired by ancient Xilinx PowerPC platforms. Here is the thread about it. https://lore.kernel.org/linux-next/48d3232d-0f1d-42ea-3109-f44bbabfa2e8@xil… I have been talking about it internally and there is no interest in these platforms and it is also orphan for quite a long time. None is really running/testing these platforms regularly that's why I think it makes sense to remove them also with drivers which are specific to this platform. U-Boot support was removed in 2017 without anybody complain about it https://github.com/Xilinx/u-boot-xlnx/commit/98f705c9cefdfdba62c069821bbba1… Based on current ppc/next. If anyone has any objection about it, please let me know. Thanks, Michal Michal Simek (2): sound: ac97: Remove sound driver for ancient platform powerpc: Remove Xilinx PPC405/PPC440 support Documentation/devicetree/bindings/xilinx.txt | 143 -- Documentation/powerpc/bootwrapper.rst | 28 +- MAINTAINERS | 6 - arch/powerpc/Kconfig.debug | 2 +- arch/powerpc/boot/Makefile | 7 +- arch/powerpc/boot/dts/Makefile | 1 - arch/powerpc/boot/dts/virtex440-ml507.dts | 406 ------ arch/powerpc/boot/dts/virtex440-ml510.dts | 466 ------- arch/powerpc/boot/ops.h | 1 - arch/powerpc/boot/serial.c | 5 - arch/powerpc/boot/uartlite.c | 79 -- arch/powerpc/boot/virtex.c | 97 -- arch/powerpc/boot/virtex405-head.S | 31 - arch/powerpc/boot/wrapper | 8 - arch/powerpc/configs/40x/virtex_defconfig | 75 - arch/powerpc/configs/44x/virtex5_defconfig | 74 - arch/powerpc/configs/ppc40x_defconfig | 8 - arch/powerpc/configs/ppc44x_defconfig | 8 - arch/powerpc/include/asm/xilinx_intc.h | 16 - arch/powerpc/include/asm/xilinx_pci.h | 21 - arch/powerpc/kernel/cputable.c | 39 - arch/powerpc/platforms/40x/Kconfig | 31 - arch/powerpc/platforms/40x/Makefile | 1 - arch/powerpc/platforms/40x/virtex.c | 54 - arch/powerpc/platforms/44x/Kconfig | 37 - arch/powerpc/platforms/44x/Makefile | 2 - arch/powerpc/platforms/44x/virtex.c | 60 - arch/powerpc/platforms/44x/virtex_ml510.c | 30 - arch/powerpc/platforms/Kconfig | 4 - arch/powerpc/sysdev/Makefile | 2 - arch/powerpc/sysdev/xilinx_intc.c | 88 -- arch/powerpc/sysdev/xilinx_pci.c | 132 -- arch/powerpc/xmon/ppc-dis.c | 6 - arch/powerpc/xmon/ppc-opc.c | 23 - arch/powerpc/xmon/ppc.h | 5 - drivers/char/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 2 +- sound/drivers/Kconfig | 12 - sound/drivers/Makefile | 2 - sound/drivers/ml403-ac97cr.c | 1298 ------------------ 40 files changed, 7 insertions(+), 3305 deletions(-) delete mode 100644 arch/powerpc/boot/dts/virtex440-ml507.dts delete mode 100644 arch/powerpc/boot/dts/virtex440-ml510.dts delete mode 100644 arch/powerpc/boot/uartlite.c delete mode 100644 arch/powerpc/boot/virtex.c delete mode 100644 arch/powerpc/boot/virtex405-head.S delete mode 100644 arch/powerpc/configs/40x/virtex_defconfig delete mode 100644 arch/powerpc/configs/44x/virtex5_defconfig delete mode 100644 arch/powerpc/include/asm/xilinx_intc.h delete mode 100644 arch/powerpc/include/asm/xilinx_pci.h delete mode 100644 arch/powerpc/platforms/40x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex.c delete mode 100644 arch/powerpc/platforms/44x/virtex_ml510.c delete mode 100644 arch/powerpc/sysdev/xilinx_intc.c delete mode 100644 arch/powerpc/sysdev/xilinx_pci.c delete mode 100644 sound/drivers/ml403-ac97cr.c -- 2.26.0

10 29

[alsa-devel] [PATCH] sound/soc: only first codec is master in multicodec setup
by Johannes Krude 12 May '20

12 May '20

When using multiple codecs, at most one codec should generate the master clock. All codecs except the first are therefore configured for slave mode. Before this patch all codecs in a multicodec setup had to be slaves. This is needed when e.g., connecting multiple sound hats for simultaneous playback to the raspberry pi I2S output and one of the sound hats generates the I2S clocks (https://github.com/raspberrypi/linux/pull/3337). I checked the raspberry pi kernel tree for multicodec usage with `fgrep num_codecs -R sound/soc/` and verified that all existing multicodec drivers with hardcoded format have indeed configured all codecs for slave mode. Doing a similar check on the current for-5.5 tree is more difficult since .num_codecs is now hidden behind a preprocessor macro. Signed-off-by: Johannes Krude <johannes(a)krude.de> --- sound/soc/soc-core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 058e038df..cb8952527 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1688,6 +1688,14 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, for_each_rtd_codec_dai(rtd, i, codec_dai) { + unsigned int codec_dai_fmt = dai_fmt; + - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + // there can only be one master when using multiple codecs + if (i && (codec_dai_fmt & SND_SOC_DAIFMT_MASTER_MASK)) { + codec_dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + codec_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + ret = snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); -- 2.17.1

2 2

snd_dice: Clicking artifacts with TC StudioKonnekt 48
by Mathias Buhr 10 May '20

10 May '20

Hi, thanks to this group, both of my firewire interfaces are (almost) working! Big thank you! While the TC Konnekt 24D works fine (playback and recording), the StudioKonnekt 48 produces clicking artifacts during playback when using snd_dice. This interface is working flawless on Windows and on a Jack/FFADO combination. This artifact occurs in all use cases (alsa via aplay, pulseaudio and jack) and does not seem to be in recorded streams. After recording the playback with another device, it looks like the level of the artifact is scaling with the signal and its interval is interestingly 256ms. Is there anything I can do to further debug this issue? Capture firewire packets? I would love to get this device fully working. I'm using kernel version 5.5.4 but this issue has been there for a long time. Thanks, Mathias

3 9

[alsa-devel] [PATCH v4 00/12] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 08 May '20

08 May '20

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. gpio controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. Thanks, srini Changes since v3: - Added new SLIMBus common bindings suggested by Rob. - renamed microvolts to microvolt in bindings as suggested by Rob - dumped pinctrl driver and added new simple gpio driver - few cosmetic cleanups in mfd driver. Srinivas Kandagatla (12): dt-bindings: SLIMBus: add slim devices optional properties ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: gpio: wcd934x: Add bindings for gpio gpio: wcd934x: Add support to wcd934x gpio controller ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga .../bindings/gpio/qcom,wcd934x-gpio.yaml | 47 + .../devicetree/bindings/slimbus/bus.txt | 10 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 163 + drivers/gpio/Kconfig | 8 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-wcd934x.c | 104 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + include/linux/mfd/wcd934x/registers.h | 531 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 16 files changed, 6399 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/qcom,wcd934x-gpio.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/gpio/gpio-wcd934x.c create mode 100644 drivers/mfd/wcd934x.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

4 17

[alsa-devel] [RESEND PATCH] ASoC: wm8524: Add support S32_LE
by Shengjiu Wang 05 May '20

05 May '20

Allow 32bit sample with this codec. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8524.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 91e3d1570c45..4e9ab542f648 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8524_dai_ops = { .startup = wm8524_startup, -- 2.21.0

2 1

[PATCH v4] ucm2: hdadsp: add basic ucm config
by Mateusz Gorski 04 May '20

04 May '20

Basic UCM configuration for Intel Skylake SST with HDA DSP generic machine driver enabling codec playback and capture on both HDA codec and DMIC ports. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- Changes in v4: - adjusted folder and files names to hda-dsp instead of hdadsp ucm2/hda-dsp/Hdmi1.conf | 24 ++++++++++ ucm2/hda-dsp/Hdmi2.conf | 24 ++++++++++ ucm2/hda-dsp/HiFi.conf | 96 +++++++++++++++++++++++++++++++++++++++ ucm2/hda-dsp/hda-dsp.conf | 16 +++++++ 4 files changed, 160 insertions(+) create mode 100644 ucm2/hda-dsp/Hdmi1.conf create mode 100644 ucm2/hda-dsp/Hdmi2.conf create mode 100644 ucm2/hda-dsp/HiFi.conf create mode 100644 ucm2/hda-dsp/hda-dsp.conf diff --git a/ucm2/hda-dsp/Hdmi1.conf b/ucm2/hda-dsp/Hdmi1.conf new file mode 100644 index 0000000..d402dab --- /dev/null +++ b/ucm2/hda-dsp/Hdmi1.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI1" { + Comment "HDMI/Display Port 1 Stereo" + + Value { + PlaybackPriority 300 + PlaybackPCM "hw:${CardId},10" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=17 Jack'" + } + True { + JackControl "HDMI/DP,pcm=17 Jack" + } + False { + JackControl "HDMI/DP, pcm=17 Jack" + } + } + } +} + diff --git a/ucm2/hda-dsp/Hdmi2.conf b/ucm2/hda-dsp/Hdmi2.conf new file mode 100644 index 0000000..153bbe1 --- /dev/null +++ b/ucm2/hda-dsp/Hdmi2.conf @@ -0,0 +1,24 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel cAVS platforms +# For Audio in HDA mode + +SectionDevice."HDMI2" { + Comment "HDMI/Display Port 2 Stereo" + + Value { + PlaybackPriority 400 + PlaybackPCM "hw:${CardId},11" + If.1 { + Condition { + Type ControlExists + Control "iface=CARD,name='HDMI/DP,pcm=18 Jack'" + } + True { + JackControl "HDMI/DP,pcm=18 Jack" + } + False { + JackControl "HDMI/DP, pcm=18 Jack" + } + } + } +} + diff --git a/ucm2/hda-dsp/HiFi.conf b/ucm2/hda-dsp/HiFi.conf new file mode 100644 index 0000000..cdeb215 --- /dev/null +++ b/ucm2/hda-dsp/HiFi.conf @@ -0,0 +1,96 @@ +# Usecase for stereo playback Speaker and Headset, Recording on DMIC and Headset MIC. +# For Audio in HDA mode on Intel cAVS platforms + +SectionVerb { + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + TQ "HiFi" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 1,1" + cset "name='Speaker Playback Switch' 0,0" + ] + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},7" + PlaybackMixerElem "Headphone" + PlaybackMasterElem "Master" + PlaybackVolume "Headphone Playback Volume" + PlaybackSwitch "Headphone Playback Switch" + JackControl "Front Headphone Jack" + JackHWMute "Speaker" + } +} + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='Headphone Playback Switch' 0,0" + cset "name='Speaker Playback Switch' 1,1" + ] + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId},7" + PlaybackMixerElem "Speaker" + PlaybackMasterElem "Master" + PlaybackVolume "Speaker Playback Volume" + PlaybackSwitch "Speaker Playback Switch" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + ConflictingDevice [ + "Mic" + ] + + EnableSequence [ + cset "name='media0_out mo codec0_in mi Switch' 1" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},7" + } +} + +SectionDevice."Mic" { + Comment "Digital Microphone" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},13" + } +} + +<hda-dsp/Hdmi1.conf> +<hda-dsp/Hdmi2.conf> diff --git a/ucm2/hda-dsp/hda-dsp.conf b/ucm2/hda-dsp/hda-dsp.conf new file mode 100644 index 0000000..48c07c8 --- /dev/null +++ b/ucm2/hda-dsp/hda-dsp.conf @@ -0,0 +1,16 @@ +# UCM for Intel CAVS platforms +# For Audio in HDA and DMIC mode + +Syntax 2 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} + +SectionDefaults [ + cset "name='Master Playback Switch' 1" + cset "name='codec0_out mo media0_in mi Switch' 1" + cset "name='media0_out mo codec0_in mi Switch' 1" + cset "name='mch_cap_out mo mch_cap_in mi Switch' 1" +] -- 2.17.1

3 2

Functionality of pcm_notify in snd-aloop?
by Pavel Hofman 30 Apr '20

30 Apr '20

Hi, Please how is the module params pcm_notify supposed to be used, to do what the documentation says: Break capture when PCM format/rate/channels changes? Breaking capture side operation when the playback side changes the params is very useful, but I cannot find a way to use this param properly. When the capture side is open, the playback side cannot use a different parameter than the one currently used by the capture side (the configuration space is limited) - how can the param be changed then? Thanks a lot for help. Best regards, Pavel.

4 9

[PATCH] ALSA: pcm: oss: Avoid plugin buffer overflow
by Takashi Iwai 30 Apr '20

30 Apr '20

Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 240e4702c098..c9401832967c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.16.4

2 3

[RFC PATCH v2 0/2] Introduce multi PM domains helpers
by Daniel Baluta 21 Apr '20

21 Apr '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v1: (addressed Ranjani's comments) - enhanced description for dev_multi_pm_attach return value - renamed exit_unroll_pm label to exit_detach_pm Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

4 6

[PATCH v3] ASoC: Intel: boards: split woofer and tweeter support
by mac.chiang＠intel.com 20 Apr '20

20 Apr '20

From: Mac Chiang <mac.chiang(a)intel.com> Support Woofer stereo speakers by default and optionally Tweeter stereo speakers with a DMI quirk Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - considering woofer speakers settings as general. - separate out the tweeter speakers and consistency of woofer+tweeter when hardware-specific. - check NULL condition if devm_kzalloc are failed v3: - fixed multiple alignments - add invalid value return --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 222 +++++++++++++++++++++++------ 1 file changed, 180 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index ed6c26a..2872345 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -30,6 +30,36 @@ #define CML_RT5682_CODEC_DAI "rt5682-aif1" #define NAME_SIZE 32 +#define SOF_RT1011_SPEAKER_WL BIT(0) +#define SOF_RT1011_SPEAKER_WR BIT(1) +#define SOF_RT1011_SPEAKER_TL BIT(2) +#define SOF_RT1011_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer speakers */ +static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR; + +static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1011_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1011_quirk_table[] = { + { + .callback = sof_rt1011_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Helios"), + }, + .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | + SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR), + }, + { + } +}; + static struct snd_soc_jack hdmi_jack[3]; struct hdmi_pcm { @@ -48,15 +78,16 @@ struct card_private { static const struct snd_kcontrol_new cml_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), }; +static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), SND_SOC_DAPM_SPK("WL Ext Spk", NULL), SND_SOC_DAPM_SPK("WR Ext Spk", NULL), SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -64,10 +95,13 @@ static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; +static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { - /*speaker*/ - {"TL Ext Spk", NULL, "TL SPO"}, - {"TR Ext Spk", NULL, "TR SPO"}, + /*WL/WR speaker*/ {"WL Ext Spk", NULL, "WL SPO"}, {"WR Ext Spk", NULL, "WR SPO"}, @@ -82,6 +116,12 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { {"DMic", NULL, "SoC DMIC"}, }; +static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -121,6 +161,35 @@ static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; }; +static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, + ARRAY_SIZE(cml_rt1011_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1011_tt_widgets, + ARRAY_SIZE(cml_rt1011_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, + ARRAY_SIZE(cml_rt1011_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -191,30 +260,38 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, * The feedback is captured for each codec individually. * Hence all 4 codecs use 1 Tx slot each for feedback. */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL | + SOF_RT1011_SPEAKER_WR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x4, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x8, 0x2, 4, 24); + if (ret < 0) + break; + } } - /* TDM Rx slot 2 is used for Right Woofer & Tweeters pair */ - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x1, 0x1, 4, 24); + if (ret < 0) + break; + } + + if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x2, 0x2, 4, 24); + if (ret < 0) + break; + } } } if (ret < 0) @@ -302,9 +379,7 @@ SND_SOC_DAILINK_DEF(ssp1_pin, SND_SOC_DAILINK_DEF(ssp1_codec, DAILINK_COMP_ARRAY( /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); + /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); @@ -398,6 +473,7 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { .dpcm_playback = 1, .dpcm_capture = 1, /* Capture stream provides Feedback */ .no_pcm = 1, + .init = cml_rt1011_spk_init, .ops = &cml_rt1011_ops, SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), }, @@ -412,14 +488,6 @@ static struct snd_soc_codec_conf rt1011_conf[] = { .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), .name_prefix = "WR", }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, }; /* Cometlake audio machine driver for RT1011 and RT5682 */ @@ -441,10 +509,12 @@ static struct snd_soc_card snd_soc_card_cml = { static int snd_cml_rt1011_probe(struct platform_device *pdev) { + struct snd_soc_dai_link_component *rt1011_dais_components; + struct snd_soc_codec_conf *rt1011_dais_confs; struct card_private *ctx; struct snd_soc_acpi_mach *mach; const char *platform_name; - int ret; + int ret, i; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -455,6 +525,73 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; + dmi_check_system(sof_rt1011_quirk_table); + + dev_info(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); + + if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | + SOF_RT1011_SPEAKER_TR)) { + rt1011_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_confs) + return -ENOMEM; + + rt1011_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + if (!rt1011_dais_components) + return -ENOMEM; + + for (i = 0; i < SPK_CH; i++) { + rt1011_dais_confs[i].dlc.name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + + if (!rt1011_dais_confs[i].dlc.name) + return -ENOMEM; + + switch (i) { + case 0: + rt1011_dais_confs[i].name_prefix = "WL"; + break; + case 1: + rt1011_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1011_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1011_dais_confs[i].name_prefix = "TR"; + break; + default: + return -EINVAL; + } + rt1011_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1011:0%d", + i); + if (!rt1011_dais_components[i].name) + return -ENOMEM; + + rt1011_dais_components[i].dai_name = CML_RT1011_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1011_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1011_rt5682_dailink); i++) { + if (!strcmp(cml_rt1011_rt5682_dailink[i].codecs->dai_name, + CML_RT1011_CODEC_DAI)) { + cml_rt1011_rt5682_dailink[i].codecs = rt1011_dais_components; + cml_rt1011_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, platform_name); @@ -482,5 +619,6 @@ MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mo MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:cml_rt1011_rt5682"); -- 2.7.4

2 1

[PATCH 0/2] ASoC: qdsp6: fix default FE dais and routings.
by Srinivas Kandagatla 20 Apr '20

20 Apr '20

QDSP6 Frontend dais can be configured to work in rx or tx or both rx/tx mode, however the default routing do not honour this DT configuration making sound card fail to probe. FE dais are also not fully honouring device tree configuration. Fix both of them. Originally issue was reported by Vinod Koul Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing sound/soc/qcom/qdsp6/q6asm-dai.c | 30 +++++++++++++++++++++++------- sound/soc/qcom/qdsp6/q6routing.c | 19 ------------------- 2 files changed, 23 insertions(+), 26 deletions(-) -- 2.21.0

4 11

[PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Ivan Mironov 17 Apr '20

17 Apr '20

ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 89811086db1d..3f4c421b7e2c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 "Xonar U7" 1 "ASUS XONAR U5" 1 + "XONAR SOUND CARD" 1 # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 -- 2.25.1

2 1

[PATCH v2] slimbus: core: Set fwnode for a device when setting of_node
by Saravana Kannan 17 Apr '20

17 Apr '20

When setting the of_node for a newly created device, also set the fwnode. This allows fw_devlink feature to work for slimbus devices. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 526e3215d8fe..44228a5b246d 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -163,8 +163,10 @@ static int slim_add_device(struct slim_controller *ctrl, INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - if (node) + if (node) { sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); + } dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, -- 2.25.1.696.g5e7596f4ac-goog

3 3

[PATCH v1] slimbus: core: Fix mismatch in of_node_get/put
by Saravana Kannan 17 Apr '20

17 Apr '20

Also, remove some unnecessary NULL checks. The functions in question already do NULL checks. Signed-off-by: Saravana Kannan <saravanak(a)google.com> --- drivers/slimbus/core.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 44228a5b246d..ae1e248a8fb8 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -162,11 +162,8 @@ static int slim_add_device(struct slim_controller *ctrl, sbdev->ctrl = ctrl; INIT_LIST_HEAD(&sbdev->stream_list); spin_lock_init(&sbdev->stream_list_lock); - - if (node) { - sbdev->dev.of_node = of_node_get(node); - sbdev->dev.fwnode = of_fwnode_handle(node); - } + sbdev->dev.of_node = of_node_get(node); + sbdev->dev.fwnode = of_fwnode_handle(node); dev_set_name(&sbdev->dev, "%x:%x:%x:%x", sbdev->e_addr.manf_id, @@ -285,6 +282,7 @@ EXPORT_SYMBOL_GPL(slim_register_controller); /* slim_remove_device: Remove the effect of slim_add_device() */ static void slim_remove_device(struct slim_device *sbdev) { + of_node_put(sbdev->dev.of_node); device_unregister(&sbdev->dev); } -- 2.25.1.696.g5e7596f4ac-goog

2 1

[PATCH] ASoC: qcom: lpass-cpu: support full duplex operation
by Srinivas Kandagatla 16 Apr '20

16 Apr '20

From: Takahide Higuchi <takahidehiguchi(a)gmail.com> This patch fixes a bug where playback on bidirectional I2S interface stops when we start recording on the same interface. We use regmap_update_bits instead of regmap_write so that we will not clear SPKEN and SPKMODE bits when we start/stop recording. Signed-off-by: Takahide Higuchi <takahidehiguchi(a)gmail.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ sound/soc/qcom/lpass-cpu.c | 24 ++++++++++++++++++------ sound/soc/qcom/lpass-lpaif-reg.h | 2 +- 3 files changed, 21 insertions(+), 7 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 6575da549237..85079c697faa 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -121,6 +121,8 @@ static struct snd_soc_dai_driver apq8016_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .symmetric_samplebits = 1, + .symmetric_rates = 1, }, }; diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92baf3..dc8acb380b6f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -72,6 +72,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t format = params_format(params); unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); + unsigned int mask; unsigned int regval; int bitwidth, ret; @@ -81,6 +82,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return bitwidth; } + mask = LPAIF_I2SCTL_LOOPBACK_MASK | + LPAIF_I2SCTL_WSSRC_MASK | + LPAIF_I2SCTL_BITWIDTH_MASK; regval = LPAIF_I2SCTL_LOOPBACK_DISABLE | LPAIF_I2SCTL_WSSRC_INTERNAL; @@ -100,6 +104,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mask |= LPAIF_I2SCTL_SPKMODE_MASK | LPAIF_I2SCTL_SPKMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_SPKMODE_SD0; @@ -127,6 +132,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { + mask |= LPAIF_I2SCTL_MICMODE_MASK | LPAIF_I2SCTL_MICMONO_MASK; switch (channels) { case 1: regval |= LPAIF_I2SCTL_MICMODE_SD0; @@ -155,9 +161,9 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, } } - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - regval); + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, regval); if (ret) { dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); return ret; @@ -178,11 +184,17 @@ static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + unsigned int mask; int ret; - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mask = LPAIF_I2SCTL_SPKMODE_MASK; + else + mask = LPAIF_I2SCTL_MICMODE_MASK; + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), + mask, 0); if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 3d74ae123e9d..7a2b9cf99976 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -56,7 +56,7 @@ #define LPAIF_I2SCTL_MICMODE_6CH (7 << LPAIF_I2SCTL_MICMODE_SHIFT) #define LPAIF_I2SCTL_MICMODE_8CH (8 << LPAIF_I2SCTL_MICMODE_SHIFT) -#define LPAIF_I2SCTL_MIMONO_MASK GENMASK(3, 3) +#define LPAIF_I2SCTL_MICMONO_MASK GENMASK(3, 3) #define LPAIF_I2SCTL_MICMONO_SHIFT 3 #define LPAIF_I2SCTL_MICMONO_STEREO (0 << LPAIF_I2SCTL_MICMONO_SHIFT) #define LPAIF_I2SCTL_MICMONO_MONO (1 << LPAIF_I2SCTL_MICMONO_SHIFT) -- 2.21.0

3 3

[alsa-devel] [PATCH] ASoC: stm32: sai: fix master clock management
by Olivier Moysan 16 Apr '20

16 Apr '20

When master clock is used, master clock rate is set exclusively. Parent clocks of master clock cannot be changed after a call to clk_set_rate_exclusive(). So the parent clock of SAI kernel clock must be set before. Ensure also that exclusive rate operations are balanced in STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 64 +++++++++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 17 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..77aa38183955 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -70,6 +70,7 @@ #define SAI_IEC60958_STATUS_BYTES 24 #define SAI_MCLK_NAME_LEN 32 +#define SAI_RATE_11K 11025 /** * struct stm32_sai_sub_data - private data of SAI sub block (block A or B) @@ -311,6 +312,25 @@ static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, return ret; } +static int stm32_sai_set_parent_clock(struct stm32_sai_sub_data *sai, + unsigned int rate) +{ + struct platform_device *pdev = sai->pdev; + struct clk *parent_clk = sai->pdata->clk_x8k; + int ret; + + if (!(rate % SAI_RATE_11K)) + parent_clk = sai->pdata->clk_x11k; + + ret = clk_set_parent(sai->sai_ck, parent_clk); + if (ret) + dev_err(&pdev->dev, " Error %d setting sai_ck parent clock. %s", + ret, ret == -EBUSY ? + "Active stream rates conflict\n" : "\n"); + + return ret; +} + static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, unsigned long *prate) { @@ -492,25 +512,29 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - if (dir == SND_SOC_CLOCK_OUT) { + if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, (unsigned int)~SAI_XCR1_NODIV); if (ret < 0) return ret; - dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); - sai->mclk_rate = freq; + /* If master clock is used, set parent clock now */ + ret = stm32_sai_set_parent_clock(sai, freq); + if (ret) + return ret; - if (sai->sai_mclk) { - ret = clk_set_rate_exclusive(sai->sai_mclk, - sai->mclk_rate); - if (ret) { - dev_err(cpu_dai->dev, - "Could not set mclk rate\n"); - return ret; - } + ret = clk_set_rate_exclusive(sai->sai_mclk, freq); + if (ret) { + dev_err(cpu_dai->dev, + ret == -EBUSY ? + "Active streams have incompatible rates" : + "Could not set mclk rate\n"); + return ret; } + + dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq); + sai->mclk_rate = freq; } return 0; @@ -917,11 +941,13 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, int div = 0, cr1 = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); + int ret; - if (!(rate % 11025)) - clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k); - else - clk_set_parent(sai->sai_ck, sai->pdata->clk_x8k); + if (!sai->sai_mclk) { + ret = stm32_sai_set_parent_clock(sai, rate); + if (ret) + return ret; + } sai_clk_rate = clk_get_rate(sai->sai_ck); if (STM_SAI_IS_F4(sai->pdata)) { @@ -1080,9 +1106,13 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, SAI_XCR1_NODIV); - clk_disable_unprepare(sai->sai_ck); + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } - clk_rate_exclusive_put(sai->sai_mclk); + clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); sai->substream = NULL; -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: pcm: fix error handling when try_module_get() fails.
by Ranjani Sridharan 16 Apr '20

16 Apr '20

Handle error before returning when try_module_get() fails to prevent inconsistent mutex lock/unlock. Fixes: 52034add7 (ASoC: pcm: update module refcount if module_get_upon_open is set) Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d21247546f7f..be80a12fba27 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -518,8 +518,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) continue; if (component->driver->module_get_upon_open && - !try_module_get(component->dev->driver->owner)) - return -ENODEV; + !try_module_get(component->dev->driver->owner)) { + ret = -ENODEV; + goto module_err; + } ret = component->driver->ops->open(substream); if (ret < 0) { @@ -636,7 +638,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) component_err: soc_pcm_components_close(substream, component); - +module_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: -- 2.17.1

2 5

[alsa-devel] [PATCH] ASoC: wcd9335: Fix missing regmap requirement
by Marc Gonzalez 16 Apr '20

16 Apr '20

wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip' Signed-off-by: Marc Gonzalez <marc.w.gonzalez(a)free.fr> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419142111b6d..33778dc99108 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1162,6 +1162,7 @@ config SND_SOC_WCD9335 tristate "WCD9335 Codec" depends on SLIMBUS select REGMAP_SLIMBUS + select REGMAP_IRQ help The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports Qualcomm Technologies, Inc. (QTI) multimedia solutions, -- 2.17.1

3 4

monotonic raw setup seems buggy
by sylvain.bertrand＠gmail.com 15 Apr '20

15 Apr '20

Hi, I am writting an alsa application and while working on pcm status timings, I noticed that setting monotonic_raw timings in sw_params while the hardware device plugin is already running with monotonic timings did not produce an error. Follow the pcm dump which shows monotonic_raw being set on all plugin instances but the already running hw plugin instance. Is this a bug, or am I doing something the wrong way? Plug PCM: Linear Integer <-> Linear Float conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_NONINTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC_RAW period_step : 1 avail_min : 1024 period_event : 1 start_threshold : 1 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 1484839806 cheers, -- Sylvain

5 20

[PATCH v1 0/2] Support headset on Tegra boards that use WM8903
by Dmitry Osipenko 14 Apr '20

14 Apr '20

Hello, Some devices have a 4-pin jack instead of a 3-pin and currently the WM8903 configuration is hardcoded to the case of 3-pin jack in the Tegra's ASoC driver. A new device-tree property is required in order to convey that hardware has a 4-pin jack, and thus, microphone's detection needs to be done in a different way. In particular this is needed for Acer A500 tablet device that has a 4-pin headset jack, otherwise userspace sees headset instead of headphones and internal microphone isn't enabled by ALSA UCM rule when it should be. Please review and apply, thanks in advance. Dmitry Osipenko (2): dt-bindings: sound: tegra-wm8903: Document new nvidia,headset property ASoC: tegra: tegra_wm8903: Support nvidia,headset property .../devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + sound/soc/tegra/tegra_wm8903.c | 6 +++++- 2 files changed, 6 insertions(+), 1 deletion(-) -- 2.25.1

4 10

[PATCH v2 1/3] dt-bindings: sound: convert rockchip i2s bindings to yaml
by Johan Jonker 14 Apr '20

14 Apr '20

Current dts files with 'i2s' nodes are manually verified. In order to automate this process rockchip-i2s.txt has to be converted to yaml. Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changed V2: Add Reviewed-by --- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 ---------- .../devicetree/bindings/sound/rockchip-i2s.yaml | 106 +++++++++++++++++++++ 2 files changed, 106 insertions(+), 49 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt deleted file mode 100644 index 54aefab71..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ /dev/null @@ -1,49 +0,0 @@ -* Rockchip I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-i2s": for rk3066 - - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 - - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 - - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 - - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 - - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 - - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "i2s_hclk": clock for I2S BUS - - "i2s_clk" : clock for I2S controller -- rockchip,playback-channels: max playback channels, if not set, 8 channels default. -- rockchip,capture-channels: max capture channels, if not set, 2 channels default. - -Required properties for controller which support multi channels -playback/capture: - -- rockchip,grf: the phandle of the syscon node for GRF register. - -Example for rk3288 I2S controller: - -i2s@ff890000 { - compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; - reg = <0xff890000 0x10000>; - interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&pdma1 0>, <&pdma1 1>; - dma-names = "tx", "rx"; - clock-names = "i2s_hclk", "i2s_clk"; - clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; - rockchip,playback-channels = <8>; - rockchip,capture-channels = <2>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml new file mode 100644 index 000000000..eff06b4b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml @@ -0,0 +1,106 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip I2S controller + +description: + The I2S bus (Inter-IC sound bus) is a serial link for digital + audio data transfer between devices in the system. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-i2s + - items: + - enum: + - rockchip,px30-i2s + - rockchip,rk3036-i2s + - rockchip,rk3188-i2s + - rockchip,rk3228-i2s + - rockchip,rk3288-i2s + - rockchip,rk3328-i2s + - rockchip,rk3366-i2s + - rockchip,rk3368-i2s + - rockchip,rk3399-i2s + - const: rockchip,rk3066-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for I2S controller + - description: clock for I2S BUS + + clock-names: + items: + - const: i2s_clk + - const: i2s_hclk + + dmas: + items: + - description: TX DMA Channel + - description: RX DMA Channel + + dma-names: + items: + - const: tx + - const: rx + + rockchip,capture-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 2 + description: + Max capture channels, if not set, 2 channels default. + + rockchip,playback-channels: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + default: 8 + description: + Max playback channels, if not set, 8 channels default. + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property for controllers which support multi channel + playback/capture. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3288-cru.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>; + clock-names = "i2s_clk", "i2s_hclk"; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + rockchip,capture-channels = <2>; + rockchip,playback-channels = <8>; + }; -- 2.11.0

3 4

[alsa-devel] [PATCH] alsactl: avoid needless wakeups in monitor loop.
by Zev Weiss 14 Apr '20

14 Apr '20

The timeout wasn't really being used for anything; disabling it should reduce idle energy consumption slightly. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..6d9328d899a4 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -340,7 +340,7 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, int count; int i; - count = epoll_wait(epfd, epev, max_ev_count, 200); + count = epoll_wait(epfd, epev, max_ev_count, -1); if (count < 0) { err = count; break; -- 2.23.0

3 3

[alsa-devel] [PATCH] alsactl: don't exit on EINTR from epoll_wait().
by Zev Weiss 14 Apr '20

14 Apr '20

Previously, things like attaching strace to a running 'alsactl monitor' process would cause it to exit. Signed-off-by: Zev Weiss <zev(a)bewilderbeest.net> --- alsactl/monitor.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsactl/monitor.c b/alsactl/monitor.c index 6b090e4f5c92..cf4167bee165 100644 --- a/alsactl/monitor.c +++ b/alsactl/monitor.c @@ -342,6 +342,8 @@ static int run_dispatcher(int epfd, int sigfd, int infd, struct list_head *srcs, count = epoll_wait(epfd, epev, max_ev_count, 200); if (count < 0) { + if (errno == EINTR) + continue; err = count; break; } -- 2.23.0

3 2

[PATCH v2 0/6] ASoC: topology: Propagate error appropriately
by Amadeusz Sławiński 09 Apr '20

09 Apr '20

v1: Check if kstrdup succeeded. v2: Remove unneeded freeing, which is performed in another place by dobj handlers. Additionally for functions which have return status which was ignored, perform success checks and handle failures in appropriate way. Amadeusz Sławiński (6): ASoC: topology: Add missing memory checks ASoC: topology: Check return value of soc_tplg_create_tlv ASoC: topology: Check return value of soc_tplg_*_create ASoC: topology: Check soc_tplg_add_route return value ASoC: topology: Check return value of pcm_new_ver ASoC: topology: Check return value of soc_tplg_dai_config sound/soc/soc-topology.c | 113 ++++++++++++++++++++++++++++++--------- 1 file changed, 88 insertions(+), 25 deletions(-) -- 2.17.1

5 23

[PATCH 1/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 09 Apr '20

09 Apr '20

snd_soc_runtime_activate() and snd_soc_runtime_deactivate() require locked pcm_mutex. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..42d416ac7e9b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -207,7 +207,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -285,7 +287,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) else stream = SNDRV_PCM_STREAM_CAPTURE; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.21.0

3 5

TRX40 Realtek ALC1220-VB USB Audio
by laurent Trinques 07 Apr '20

07 Apr '20

Hello, sudo dmesg -lerr -lcrit [ 0.386478] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT1A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386484] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386488] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT1A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386490] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386493] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT2A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386495] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386497] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT2A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386499] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386501] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT3A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386504] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386506] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT3A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386508] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386510] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.WT4A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386513] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386516] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT4A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386518] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386520] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CA.MT5A], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386523] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386526] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT1B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386529] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386531] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT1B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386534] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386536] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT2B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386538] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386541] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT2B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386543] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386546] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT3B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386548] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386550] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT3B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386553] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386555] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.WT4B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386557] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386560] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT4B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386562] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386565] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CB.MT5B], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386567] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386571] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT1C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386573] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386575] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT1C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386578] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386580] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT2C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386583] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386585] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT2C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386587] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386590] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT3C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386592] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386595] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT3C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386597] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386599] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.WT4C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386602] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386604] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT4C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386606] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386609] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CC.MT5C], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386611] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386615] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT1D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386617] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386620] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT1D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386622] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386624] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT2D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386627] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386629] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT2D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386632] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386634] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT3D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386636] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386639] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT3D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386641] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386644] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.WT4D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386646] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386648] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT4D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386651] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 0.386653] ACPI BIOS Error (bug): Failure creating named object [\_SB.I2CD.MT5D], AE_ALREADY_EXISTS (20191018/dswload2-323) [ 0.386656] ACPI Error: AE_ALREADY_EXISTS, During name lookup/catalog (20191018/psobject-220) [ 2.527434] sp5100-tco sp5100-tco: Watchdog hardware is disabled [ 2.545332] Error: Driver 'pcspkr' is already registered, aborting... [ 2.572542] snd_hda_intel 0000:23:00.4: no codecs found! [ 8.169209] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.169214] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.430796] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.430800] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.431164] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.483290] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.483294] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.483662] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.628066] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.628071] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.628557] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.849035] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 8.849039] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 8.849405] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.040530] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.040534] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.040903] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.233150] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.233154] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.233522] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.425023] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.425027] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.425389] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.617141] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.617145] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.617636] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.809016] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 9.809020] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 9.809385] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.000383] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.000387] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.000754] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.417141] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.417146] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.417641] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.467123] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 10.467127] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 10.467870] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.148540] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.148545] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.149030] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.211031] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.211035] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.211402] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.289284] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.289289] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.289654] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.480906] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.480911] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.481275] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.673023] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.673028] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.673392] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.864772] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 14.864777] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 14.865138] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.057144] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.057149] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.057639] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.248638] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.248643] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.249010] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.440883] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.440887] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.441251] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.633004] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 15.633009] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 15.633373] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 16.016995] usb 7-5: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 16.017000] usb 7-5: 19:0: cannot get min/max values for control 12 (id 19) [ 16.017491] usb 7-5: cannot get ctl value: req = 0x81, wValue = 0xc00, wIndex = 0x1300, type = 4 Best regards, Laurent

2 4

[PATCH 0/4] ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link
by Cezary Rojewski 06 Apr '20

06 Apr '20

As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/haswell.c | 1 - 4 files changed, 4 deletions(-) -- 2.17.1

5 22

[PATCH] quirk for Pioneer DJ DJM-250MK2 usb sound card
by František Kučera 03 Apr '20

03 Apr '20

Hello, I have created a quirk that adds support for the Pioneer DJ DJM-250MK2 mixer (acts like a USB sound card, but Vendor Specific Class). Output (from computer to card) is working: 8 channels, 48 000 Hz, S24_3LE. Input is not working. I tried, but I was unable to make it working even under an officially supported OS with Pioneer drivers - Audacity or Mixxx see the channels, but there is no signal. It probably works only with Pioneer proprietary application. But I hope that even mere output support will be useful for someone. (it is usable for playback, just not for DVS) Franta Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..dcaf9eed9a41 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC

4 6

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 02 Apr '20

02 Apr '20

This commit adds alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..344b06df0e15 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3565,8 +3565,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->i2s_dev->name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3587,9 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

3 4

[PATCH] pcm: add chmap option to route plugin
by Jonas Holmberg 31 Mar '20

31 Mar '20

Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..d3e5f3ff 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

2 1

[PATCH v2 1/3] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 31 Mar '20

31 Mar '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- Changed V2: dmas and dma-names layout --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ----------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 94 ++++++++++++++++++++++ 2 files changed, 94 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..d1c72c8a5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru-common.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

4 9

[PATCH] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Stephan Gerhold 31 Mar '20

31 Mar '20

At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- I'm still quite confused about the true meaning of SNDRV_PCM_INFO_BATCH, so please correct me if I'm wrong :) The tsched=0 workaround can be found in Linaro distributions for QCOM devices for example: - https://git.linaro.org/ci/fai.git/commit/?id=63494268b654d80df033f4cdeccf8f… - https://github.com/ndechesne/meta-qcom/commit/7035dfeadd1c434fc7613730ac380… This patch allows removing that workaround since audio then works without any configuration changes. --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index f6c7cddf08e8..125af00bba53 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -78,7 +78,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -100,7 +100,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.26.0

5 4

[PATCH v2 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial state. This is to ensure that the default mixer choice will be correctly powered up during initialization. Invert flag is used to select initial state of the virtual switch. Since actual hardware can't be disconnected by virtual switch, connected is better choice as initial state in many cases. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..17962564866d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial state. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + p->connect = invert; } } -- 2.21.0

2 1

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

2 1

[PATCH] pcm: add chmap option to route plugin
by jonashg＠axis.com 31 Mar '20

31 Mar '20

From: Jonas Holmberg <jonashg(a)axis.com> Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..a7b6552c 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

2 1

[PATCH] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 31 Mar '20

31 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 78def224f59c (ALSA: hda/realtek - Add Headset Mic supported) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

2 3

[PATCH 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

3 6

[PATCH v2 0/3] Properly fix headphone noise on the XPS 13 and other ALC256 devices
by Thomas Hebb 31 Mar '20

31 Mar '20

The root cause of various pervasive audio problems on the XPS 13 9350/9360, mostly relating to the headphone jack, turns out to be an undocumented feature of the ALC256 and similar codecs that routes audio along paths not exposed in the HDA node graph. The best we've had so far to configure this feature is magic numbers provided by Realtek, none of which have fully fixed all issues. This series documents the "PC Beep Hidden Register", which controls the feature and which I've reverse engineered using black box techniques, and uses my findings to hopefully fix the headphone issues on my XPS 13 once and for all. Changes in v2: - Change fixed value from 0x4727 to 0x5757, which should behave identically, on advice from Kailang. Thomas Hebb (3): ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256 ALSA: hda/realtek - Set principled PC Beep configuration for ALC256 ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 2 - .../sound/hd-audio/realtek-pc-beep.rst | 129 ++++++++++++++++++ sound/pci/hda/patch_realtek.c | 49 ++----- 4 files changed, 139 insertions(+), 42 deletions(-) create mode 100644 Documentation/sound/hd-audio/realtek-pc-beep.rst -- 2.25.2

3 6

[PATCH 0/5] Add support for SOF on i.MX8M
by Daniel Baluta 31 Mar '20

31 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for SOF on i.MX8M family. First board from this family that has a DSP is i.MX8MP. First 2 patches are trying to fix some compilation issues, the next two are adding the imx8m support and the last one adds the devicetree binding. Daniel Baluta (4): ASoC: SOF: imx: fix undefined reference issue ASoC: SOF: imx: Add i.MX8M HW support ASoC: SOF: Add i.MX8MP device descriptor dt-bindings: dsp: fsl: Add fsl,imx8mp-dsp entry YueHaibing (1): ASoC: SOF: imx8: Fix randbuild error .../devicetree/bindings/dsp/fsl,dsp.yaml | 2 + sound/soc/sof/imx/Kconfig | 32 +- sound/soc/sof/imx/Makefile | 2 + sound/soc/sof/imx/imx8m.c | 279 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 14 + 5 files changed, 325 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx8m.c -- 2.17.1

4 9

PATCH: ALSA USB MIDI: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Andreas Steinmetz 31 Mar '20

31 Mar '20

The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> --- a/sourd/usb/midi.c 2020-03-26 17:34:40.073929497 +0100 +++ b/sound/usb/midi.c 2020-03-26 17:43:39.123252612 +0100 @@ -1839,7 +1839,8 @@ struct usb_host_endpoint *hostep; struct usb_endpoint_descriptor *ep; struct usb_ms_endpoint_descriptor *ms_ep; - int i, epidx; + int i, epidx, extralen; + unsigned char *extra; intf = umidi->iface; if (!intf) @@ -1863,12 +1864,21 @@ ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || + extralen = hostep->extralen; + extra = hostep->extra; +repeat: ms_ep = (struct usb_ms_endpoint_descriptor *)extra; + if (extralen < 1) + continue; + if (extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) - continue; + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) { + if (!extra[0]) + continue; + extralen -= extra[0]; + extra += extra[0]; + goto repeat; + } if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) {

2 3

[PATCH] ASoC: snd_soc_dai_set_fmt add substream independence.
by Matt Flax 31 Mar '20

31 Mar '20

This patch is aims to start a stronger discussion on allowing both CPU and Codec dais to set formats independently based on direction. Currently it is very difficult to change stream formats because there is no way to specify it independently. A previous discussion w.r.t. a codec's stream dependent format setting required a codec's dai to be split. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/164492.html A previous discussion w.r.t. changing format based on the stream direction w.r.t. a CPU's dai showed the difficulty in setting fmt based on direection in thw hw_params function. See here : https://mailman.alsa-project.org/pipermail/alsa-devel/2020-March/165101.html It seems that it may be necessary to allow the stream direction as an input argument to snd_soc_dai_set_fmt. This patch is not complete, as there are many codecs and cpus which would require this change. However this patch is aimed as a discussion point. One example of a sound card which requires independent stream formats is an isolated sound card, such as the Audio Injector Isolated sound card. The magnetic isolation chips on the sound card require stream fomats to be different because of digital latency variations on the I2S lines. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- include/sound/soc-dai.h | 4 ++-- sound/soc/soc-core.c | 5 +++-- sound/soc/soc-dai.c | 4 ++-- 3 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index eaaeb00e9e84..5071e006389b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -123,7 +123,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); @@ -186,7 +186,7 @@ struct snd_soc_dai_ops { * DAI format configuration * Called by soc_card drivers, normally in their hw_params. */ - int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt, int stream); int (*xlate_tdm_slot_mask)(unsigned int slots, unsigned int *tx_mask, unsigned int *rx_mask); int (*set_tdm_slot)(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..27a6f01dc2d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1474,10 +1474,11 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; unsigned int i; - int ret; + int ret, stream; for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + int stream = 0; + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt, stream); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 51031e330179..1f8ee7875656 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -94,11 +94,11 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); * * Configures the DAI hardware format and clocking. */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, int stream) { if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; - return dai->driver->ops->set_fmt(dai, fmt); + return dai->driver->ops->set_fmt(dai, fmt, stream); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); -- 2.20.1

2 6

[PATCH 3/3] ASoC: fix regwmask
by 이경택 31 Mar '20

31 Mar '20

If regwshift is 32 and the selected architecture compiles '<<' operator for signed int literal into rotating shift, '1<<regwshift' became 1 and it makes regwmask to 0x0. The literal is set to unsigned long to get intended regwmask. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 652657dc6809..55ffb34be95e 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -825,7 +825,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long min = mc->min; @@ -874,7 +874,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long max = mc->max; -- 2.21.0

2 5

[PATCH v5 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 31 Mar '20

31 Mar '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 5 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 20 +- sound/soc/fsl/fsl_asrc.c | 303 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2111 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3203 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

5 20

[PATCH] Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
by Takashi Iwai 31 Mar '20

31 Mar '20

This reverts commit 645c08f17f477915f6d900b767e789852f150054 which was reported to break the build a program using this header. The original issue was addressed in the alsa-lib side recently, so we can make the header more self-contained again. Reported-by: Dmitry V. Levin <ldv(a)altlinux.org> Fixes: 645c08f17f47 ("ALSA: uapi: Drop asound.h inclusion from asoc.h") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6048553c119d..a74ca232f1fc 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -17,6 +17,7 @@ #define __LINUX_UAPI_SND_ASOC_H #include <linux/types.h> +#include <sound/asound.h> /* * Maximum number of channels topology kcontrol can represent. -- 2.16.4

1 0

[PATCH 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial value 0. This is to ensure that the default mixer choice will be correctly powered up during initialization. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..fc55a0534252 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,16 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial value 0. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + val = 0; + if (invert) + val = max - val; + p->connect = !!val; } } -- 2.21.0

3 7

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

1 0

[PATCH 0/3] fixes for topology
by 이경택 31 Mar '20

31 Mar '20

Hi, I'd like to submit couple of patches which will fix few problems in corner case scenarios using topology. Any feedback will be greatly appreciated. Lee Gyeongtaek Lee (3): ASoC: dapm: connect virtual mux with default value ASoC: topology: use name_prefix for new kcontrol ASoC: fix regwmask sound/soc/soc-dapm.c | 11 ++++++++++- sound/soc/soc-ops.c | 4 ++-- sound/soc/soc-topology.c | 2 +- 3 files changed, 13 insertions(+), 4 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 2

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

1 0

[PATCH v3] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 31 Mar '20

31 Mar '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. This patch is assuming that "sound-dai" is already defined. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - uses maxItems for many place which were already defined. - assuming that "sound-dai" is already defined - (@.*) -> (@[0-9a-f]+) .../devicetree/bindings/sound/simple-card.txt | 351 ------------- .../bindings/sound/simple-card.yaml | 487 ++++++++++++++++++ 2 files changed, 487 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..cd5ea50daa22 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,487 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + bitclock-master: + description: Indicates dai-link bit clock master + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - maxItems: 1 + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + maxItems: 1 + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + maxItems: 1 + + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + $ref: "#/definitions/clocks" + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": + const: 1 + "#size-cells": + const: 0 + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + +# use patternProperties to use "#definitions/xxx" +patternProperties: + "^simple-audio-card,widgets$": + $ref: "#/definitions/widgets" + "^simple-audio-card,routing$": + $ref: "#/definitions/routing" + "^simple-audio-card,cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + "^simple-audio-card,hp-det-gpio$": + maxItems: 1 + "^simple-audio-card,mic-det-gpio$": + maxItems: 1 + + "^simple-audio-card,dai-link(@[0-9a-f]+)?$": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + maxItems: 1 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + + patternProperties: + "^cpu(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + "^codec(@[0-9a-f]+)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

2 2

[alsa-devel] [PATCH v3 0/7] ASoC: soc-pcm cleanup step2
by Kuninori Morimoto 31 Mar '20

31 Mar '20

Hi Mark These are v3 of soc-pcm cleanup. Original [5/7] [6/7] patch call open / start function for all DAI / component. And, it calls close / stop function for all if some of them got error. But in such case, error occured DAI / component don't need to call close / stop. This issue can be solved if it had flag, and my local tree has such patch. But it was planed to post to ML a little later. This time, I merged original patch and this new flag patch on [5/7] [6/7]. These mention that "it is prepare for soc-pcm-open() cleanup". But it will be happen later. These are based on v5.6-rc1 Kuninori Morimoto (8): 1) ASoC: soc-pcm: add snd_soc_runtime_action() 2) ASoC: soc-pcm: adjustment for DAI member 0 reset 3) ASoC: soc-pcm: add for_each_dapm_widgets() macro 4) ASoC: soc-pcm: don't use bit-OR'ed error 5) ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once 6) ASoC: soc-pcm: call snd_soc_component_open/close() once 7) ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() 8) ASoC: soc-pcm: tidyup soc_pcm_open() order include/sound/soc-component.h | 7 +- include/sound/soc-dai.h | 5 +- include/sound/soc-dapm.h | 5 + sound/soc/soc-component.c | 35 ++++-- sound/soc/soc-dai.c | 11 +- sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 246 +++++++++++++++++++----------------------- 7 files changed, 161 insertions(+), 156 deletions(-) -- 2.7.4

6 32

[alsa-devel] [PATCH 0/5] ALSA: Fix uapi headers
by Takashi Iwai 30 Mar '20

30 Mar '20

Hi, here is a series of fixes for include/uapi/sound/*.h files and the relevant code changes, so that they can be used for building alsa-tools programs again. The corresponding alsa-lib update will be followed. Takashi === Takashi Iwai (5): ALSA: emu10k1: Make uapi/emu10k1.h compilable again ALSA: hdsp: Make uapi/hdsp.h compilable again ALSA: hdspm: Drop linux/types.h inclusion in uapi header ALSA: uapi: Fix typos and header inclusion in asound.h ALSA: uapi: Drop asound.h inclusion from asoc.h include/uapi/sound/asoc.h | 1 - include/uapi/sound/asound.h | 6 +++--- include/uapi/sound/emu10k1.h | 38 ++++++++++++++++++++++++++------------ include/uapi/sound/hdsp.h | 4 +--- include/uapi/sound/hdspm.h | 2 -- sound/pci/emu10k1/emufx.c | 26 ++++++++++++++------------ sound/pci/rme9652/hdsp.c | 2 +- 7 files changed, 45 insertions(+), 34 deletions(-) -- 2.16.4

2 7

[PATCH] ASoC: soc-dai: fix DAI startup/shutdown sequence
by Pierre-Louis Bossart 30 Mar '20

30 Mar '20

The addition of a single flag to track the DAI status prevents the DAI startup sequence from being called on capture if the DAI is already used for playback. Fix by extending the existing code with one flag per direction. Reported-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Tested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Fixes: b56be800f1292 ("ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dai.h | 2 +- sound/soc/soc-dai.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 78bac995db15..d4825b82c7a3 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,7 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started:1; + unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 19142f6e533c..8f3cad8db89a 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,12 +295,12 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (!dai->started && + if (!dai->started[substream->stream] && dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); if (ret == 0) - dai->started = 1; + dai->started[substream->stream] = 1; return ret; } @@ -308,11 +308,11 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->started && + if (dai->started[substream->stream] && dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); - dai->started = 0; + dai->started[substream->stream] = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, -- 2.20.1

2 1

snd_hda_intel/sst-acpi sound breakage on suspend/resume since 5.6-rc1
by Dominik Brodowski 30 Mar '20

30 Mar '20

Hi! While 5.5.x works fine, mainline as of ac309e7744be (v5.6-rc6+) causes me some sound-related trouble: after boot, the sound works fine -- but once I suspend and resume my broadwell-based XPS13, I need to switch to headphone and back to speaker to hear something. But what I hear isn't music but garbled output. A few dmesg snippets from v5.6-rc6-9-gac309e7744be which might be of interest. I've highlighted the lines differing from v.5.5.x which might be of special interest: ... snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) usbcore: registered new interface driver snd-usb-audio snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops) input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input13 input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input14 input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input15 input: HDA Intel HDMI HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:03.0/sound/card0/input16 input: HDA Intel HDMI HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:03.0/sound/card0/input17 Console: switching to colour frame buffer device 240x67 !!! sst-acpi INT3438:00: WARN: Device release is not defined so it is not safe to unbind this driver while in use i915 0000:00:02.0: fb0: i915drmfb frame buffer device sst-acpi INT3438:00: DesignWare DMA Controller, 8 channels psmouse serio1: synaptics: Unable to query device: -5 haswell-pcm-audio haswell-pcm-audio: Direct firmware load for intel/IntcPP01.bin failed with error -2 haswell-pcm-audio haswell-pcm-audio: fw image intel/IntcPP01.bin not available(-2) haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 rt286 i2c-INT343A:00: ASoC: sink widget DMIC1 overwritten rt286 i2c-INT343A:00: ASoC: source widget DMIC1 overwritten broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> System Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload0 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Offload1 Pin mapping ok broadwell-audio broadwell-audio: snd-soc-dummy-dai <-> Loopback Pin mapping ok broadwell-audio broadwell-audio: rt286-aif1 <-> snd-soc-dummy-dai mapping ok input: broadwell-rt286 Headset as /devices/pci0000:00/INT3438:00/broadwell-audio/sound/card1/input18 ... ALSA device list: #0: HDA Intel HDMI at 0xf7218000 irq 48 #1: DellInc.-XPS139343--0TM99H ... !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to reset, ignore it. !!! haswell-pcm-audio haswell-pcm-audio: warning: stream is NULL, no stream to free, ignore it. (these last two messages already are printed a couple of time after boot, and then again during a suspend/resume cycle. On v.5.5.y, there are similar messages "no context buffer need to restore!"). Everything is built-in, no modules are loaded. Unfortunately, I cannot bisect this issue easily -- i915 was broken for quite some time on this system[*], prohibiting boot... Thanks for taking a look at this issue! Dominik [*] https://gitlab.freedesktop.org/drm/intel/issues/1151

8 38

[PATCH] dt-bindings: sound: cs42l51: Remove unneeded I2C unit name
by Fabio Estevam 30 Mar '20

30 Mar '20

The following warning is seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/cirrus,cs42l51.example.dts:18.15-34.11: Warning (unit_address_vs_reg): /example-0/i2c@0: node has a unit name, but no reg or ranges property Fix it by removing the unneeded i2c unit name. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml index efce847a3408..83f44f07ac3f 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml @@ -49,7 +49,7 @@ required: examples: - | #include <dt-bindings/gpio/gpio.h> - i2c@0 { + i2c { #address-cells = <1>; #size-cells = <0>; -- 2.17.1

3 2

[PATCH] ASoC: bcm: Fix pointer cast warning
by Takashi Iwai 30 Mar '20

30 Mar '20

The NULL check can be done gracefully without cast. It fixes a compile warning like: sound/soc/bcm/bcm63xx-pcm-whistler.c:184:6: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] Fixes: 88eb404ccc3e ("ASoC: brcm: Add DSL/PON SoC audio driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/bcm/bcm63xx-pcm-whistler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index e46c390683e7..b7a1efc7406e 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -181,7 +181,7 @@ bcm63xx_pcm_pointer(struct snd_soc_component *component, snd_pcm_uframes_t x; struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; - if ((void *)prtd->dma_addr_next == NULL) + if (!prtd->dma_addr_next) prtd->dma_addr_next = substream->runtime->dma_addr; x = bytes_to_frames(substream->runtime, -- 2.16.4

2 1

[PATCH] ASoC: AMD: Clear format bits before setting them
by Akshu Agrawal 30 Mar '20

30 Mar '20

This avoids residual bit form previous format when the format is changed. Hence, the resultant format is not an invalid one. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 1 + sound/soc/amd/raven/acp3x.h | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 3a3c47e820ab..f160d35a6832 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -139,6 +139,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); } val = rv_readl(rtd->acp3x_base + reg_val); + val &= ~ACP3x_ITER_IRER_SAMP_LEN_MASK; val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); return 0; diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 21e7ac017f2b..03fe93913e12 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -76,6 +76,8 @@ #define ACP_POWERED_OFF 0x02 #define ACP_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_ITER_IRER_SAMP_LEN_MASK 0x38 + struct acp3x_platform_info { u16 play_i2s_instance; u16 cap_i2s_instance; -- 2.20.1

2 1

[PATCH 0/4] ucm2: Add support to SDM845 based platforms
by Srinivas Kandagatla 30 Mar '20

30 Mar '20

This patchset adds UCM support for WSA881x, WCD934x codec and two platforms, DB845c DragonBoard and Lenovo YOGA C360 Laptop that use these codecs. Tested on Ubuntu 20.04 with pulseaudio 13 along with with this fix https://github.com/pulseaudio/pulseaudio/commit/cabd387e26b5fe03661f4b89482… Thanks, srini Srinivas Kandagatla (4): ucm2: codecs: wcd934x: Add Codec sequences. ucm2: codecs: wsa881x: add codec sequences ucm2: DB845c: Add ucm for DB845c board ucm2: Add ucm for Lenovo-YOGA-C630-13Q50 laptop ucm2/DB845c/DB845c.conf | 11 +++ ucm2/DB845c/HDMI.conf | 33 +++++++ ucm2/DB845c/HiFi.conf | 46 +++++++++ ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf | 98 +++++++++++++++++++ .../Lenovo-YOGA-C630-13Q50.conf | 6 ++ ucm2/codecs/wcd934x/DefaultDisableSeq.conf | 5 + ucm2/codecs/wcd934x/DefaultEnableSeq.conf | 14 +++ ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf | 6 ++ ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf | 6 ++ .../wcd934x/HeadphoneMicDisableSeq.conf | 2 + .../codecs/wcd934x/HeadphoneMicEnableSeq.conf | 3 + ucm2/codecs/wcd934x/SpeakerDisableSeq.conf | 4 + ucm2/codecs/wcd934x/SpeakerEnableSeq.conf | 4 + ucm2/codecs/wsa881x/DefaultEnableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerDisableSeq.conf | 8 ++ ucm2/codecs/wsa881x/SpeakerEnableSeq.conf | 8 ++ 16 files changed, 262 insertions(+) create mode 100644 ucm2/DB845c/DB845c.conf create mode 100644 ucm2/DB845c/HDMI.conf create mode 100644 ucm2/DB845c/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/HiFi.conf create mode 100644 ucm2/Lenovo-YOGA-C630-13Q50/Lenovo-YOGA-C630-13Q50.conf create mode 100644 ucm2/codecs/wcd934x/DefaultDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/HeadphoneMicEnableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wcd934x/SpeakerEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/DefaultEnableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerDisableSeq.conf create mode 100644 ucm2/codecs/wsa881x/SpeakerEnableSeq.conf -- 2.21.0

2 7

[GIT PULL] ASoC updates for v5.7
by Mark Brown 30 Mar '20

30 Mar '20

The following changes since commit 16fbf79b0f83bc752cee8589279f1ebfe57b3b6e: Linux 5.6-rc7 (2020-03-22 18:31:56 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.7 for you to fetch changes up to 1c521d7e62262793789845989edca57dea24eb7d: Merge branch 'asoc-5.7' into asoc-next (2020-03-27 17:29:20 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.7 This is a very big update for the core since Morimoto-san has been rather busy continuing his refactorings to clean up a lot of the cruft that we have accumilated over the years. We've also gained several new drivers, including initial (but still not complete) parts of the Intel SoundWire support. - Lots of refactorings to modernize the code from Morimoto-san. - Conversion of SND_SOC_ALL_CODECS to use imply from Geert Uytterhoeven. - Continued refactoring and fixing of the Intel support. - Soundwire and more advanced clocking support for Realtek RT5682. - Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563 and TLV320ADCX140. ---------------------------------------------------------------- Akshu Agrawal (4): ASoC: amd: Add machine driver for Raven based platform ASoC: amd: Allow I2S wake event after ACP is powerd On ASoc: amd: Add DMIC switch capability to machine driver ASoC: amd: Fix compile warning of argument type Amadeusz Sławiński (1): ASoC: topology: Perform component check upfront Amery Song (2): ASoC: SOF: Intel: hda: remove unnecessary ROM IPC filter function ASoC: SOF: Intel: remove unnecessary waitq before loading firmware Baolin Wang (1): ASoC: sprd: Allow the MCDT driver to build into modules Bard Liao (5): ASoC: Return error if the function does not support multi-cpu ASoC: pcm: check if cpu-dai supports a given stream ASoC: SOF: Intel: hda-dai: add stream capability ASoC: SOF: topology: connect dai widget to all cpu-dais ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Benjamin Gaignard (1): ASoC: Convert cirrus,cs42l51 to json-schema Cezary Rojewski (20): ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Expose SDnFMT helpers ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs ASoC: SOF: Fix probe point getter ASoC: Intel: Skylake: Remove superfluous chip initialization ASoC: Intel: Skylake: Select hda configuration permissively ASoC: Intel: Skylake: Enable codec wakeup during chip init ASoC: Intel: Skylake: Shield against no-NHLT configurations ASoC: Intel: Allow for ROM init retry on CNL platforms ASoC: Intel: Skylake: Await purge request ack on CNL ASoC: SOF: Intel: Fix stream cleanup on hw free ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components ASoC: Intel: bdw-rt5677: Revert SSP0 link to use dummy components ASoC: Intel: bdw-rt5650: Revert SSP0 link to use dummy components Charles Keepax (1): ASoC: samsung: Update dependencies for Arizona machine drivers Colin Ian King (6): ASoC: Intel: mrfld: return error codes when an error occurs ASoC: rt5659: remove redundant assignment to variable idx ASoC: ti: davinci-mcasp: remove redundant assignment to variable ret ASoC: Intel: mrfld: fix incorrect check on p->sink ASoC: mchp-i2s-mcc: make signed 1 bit bitfields unsigned ASoC: amd: acp3x-pcm-dma: clean up two indentation issues Curtis Malainey (1): ASoC: Intel: Make glk+rt5682 echo ref dynamic Dafna Hirschfeld (1): dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format. Dan Murphy (10): ASoC: tas2562: Add support for ISENSE and VSENSE dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family ASoC: tas2562: Add support for digital volume control ASoC: tlv320adcx140: Add DRE and AGC support ASoC: tlv320adcx140: Add decimation filter support ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding ASoC: tas2562: Add entries for the TAS2563 audio amplifier ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification ASoC: tlv320adcx140: Remove undocumented property Derek Fang (3): ASoC: rl6231: Add new supports on rl6231 ASoC: rt5682: Add CCF usage for providing I2S clks ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage Dmitry Osipenko (2): ASoC: tegra: tegra_wm8903: Support DAPM events for built-in microphone ASoC: tegra-wm8903: Document built-in microphone audio source Enric Balletbo i Serra (1): ASoC: amd: AMD RV RT5682 should depends on CROS_EC Geert Uytterhoeven (7): spi: pxa2xx: Enable support for compile-testing ASoC: Use imply for SND_SOC_ALL_CODECS ASoC: sh: fsi: Restore devm_ioremap() alignment ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout ASoC: Fix SND_SOC_ALL_CODECS imply I2C fallout ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout Guennadi Liakhovetski (7): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: export DPCM runtime update functions ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load() ASoC: SOF: fix uninitialised "work" with VirtIO Gustavo A. R. Silva (1): ASoC: wm0010: Replace zero-length array with flexible-array member Jack Yu (2): ASoC: rt1015: fix typo for bypass boost control ASoC: rt1015: modify some structure to be static. James Schulman (1): MAINTAINERS: Update Cirrus Logic codec driver maintainers Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Jeff Chang (1): ASoC: MT6660 update to 1.0.8_G Jerome Brunet (23): ASoC: core: allow a dt node to provide several components ASoC: meson: g12a: extract codec-to-codec utils ASoC: meson: aiu: add audio output dt-bindings ASoC: meson: aiu: add i2s and spdif support ASoC: meson: aiu: add hdmi codec control support ASoC: meson: aiu: add internal dac codec control support ASoC: meson: axg: extract sound card utils ASoC: meson: gx: add sound card dt-binding documentation ASoC: meson: gx: add sound card support ASoC: core: ensure component names are unique ASoC: meson: aiu: remove unused encoder structure ASoC: meson: aiu: fix clk bulk size allocation ASoC: meson: aiu: fix irq registration ASoC: meson: aiu: fix acodec dai input name init ASoC: meson: codec-glue: fix pcm format cast warning ASoC: meson: aiu: simplify component addition ASoC: fix card registration regression. ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver ASoC: dpcm: remove confusing trace in dpcm_get_be() ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support Johan Jonker (2): dt-bindings: sound: convert rockchip i2s bindings to yaml dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property Jonghwan Choi (1): ASoC: tas2562: Fixed incorrect amp_level setting. Kai Vehmanen (10): ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver ASoC: SOF: Intel: hda: allow operation without i915 gfx ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close() MAINTAINERS: add entry for Sound Open Firmware drivers ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC ASoC: SOF: Intel: hda: do not leave clock gating off upon error ASoC: SOF: Intel: hda: call codec wake at chip init Karol Trzcinski (2): ASoC: SOF: Make sof_ipc_ext_data enum more rigid ASoC: SOF: Remove SOF_IPC_EXT_DMA_BUFFER Kevin Li (2): ASoC: brcm: Add DSL/PON SoC audio driver ASoC: brcm: DSL/PON SoC device tree bindings of audio driver Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Kuninori Morimoto (95): ASoC: soc-pcm: add snd_soc_runtime_action() ASoC: soc-pcm: adjustment for DAI member 0 reset ASoC: soc-pcm: add for_each_dapm_widgets() macro ASoC: soc-pcm: don't use bit-OR'ed error ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once ASoC: soc-pcm: call snd_soc_component_open/close() once ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() ASoC: soc-pcm: tidyup soc_pcm_open() order ASoC: soc-pcm: move dai_get_widget() ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code ASoC: soc-pcm: use goto and remove multi return ASoC: soc-pcm: merge playback/cature_active into stream_active ALSA: pcm.h: add for_each_pcm_streams() ASoC: soc-core: use for_each_pcm_streams() macro ASoC: soc-pcm: use for_each_pcm_streams() macro ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro ASoC: sof: sof-audio: use for_each_pcm_streams() macro ALSA: usx2y: use for_each_pcm_streams() macro ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() ASoC: soc-pcm: cleanup soc_pcm_apply_msb() ASoC: soc-pcm: add snd_soc_dai_get_widget() ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() ASoC: soc-dapm: don't use rtd->cpu_dai on for_each_rtd_cpu_dai() ASoC: soc-pcm: use defined stream ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() ASoC: soc-pcm: move dpcm_fe_dai_close() ASoC: soc-pcm: add dpcm_fe_dai_cleanup() ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() ASoC: soc-pcm: check DAI's activity more simply ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. ASoC: simple-card-utils: use for_each_pcm_streams() ASoC: soc.h: add for_each_rtd_codecs/cpus_dai() macro ASoC: Intel: use for_each_rtd_codecs/cpus_dai() macro ASoC: mediatek: use for_each_rtd_codecs/cpus_dai() macro ASoC: meson: use for_each_rtd_codecs/cpus_dai() macro ASoC: qcom: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc: use for_each_rtd_codecs/cpus_dai() macro ASoC: soc.h: remove non plural form for_each_xxx macro ASoC: soc-dapm: add for_each_card_dapms() macro ASoC: soc-dapm: add for_each_card_widgets() macro ASoC: soc-core: Merge CPU/Codec DAIs ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais() ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new() ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer Lukas Bulwahn (1): ASoC: MT6660: make spdxcheck.py happy Marek Szyprowski (1): ASoC: samsung: Silence warnings during deferred probe Mark Brown (37): Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 ASoC: soc-pcm cleanup step4 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 SoundWire: ASoC interfaces for multi-cpu dais and DisCo helpers ASoC: Add Multi CPU DAI support Merge series "ASoC: SOF: updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: wcd934x: minor fixes" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "simple-audio-card codec2codec support" from Samuel Holland <samuel(a)sholland.org>: Merge series "ASoC: soc-pcm cleanup step5" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-pcm cleanup step6" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rt5682: fix Kconfig/compilation issues when I2C is not selected" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge tag 'v5.6-rc5' into asoc-5.7 Merge series "ASoC: qdsp6: fix default FE dais and routings." from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: SOF: preparatory patches" from Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com>: Merge series "ASoC: brcm: add dsl and pon chip audio driver" from Kevin Li <kevin-ke.li(a)broadcom.com>: Merge series "ASoC: Intel: machine driver updates for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: multi-cpu dais, IPC and Intel improvements for 5.7" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ALSA: compress: Add wma, alac and ape support" from Vinod Koul <vkoul(a)kernel.org>: Merge series "ASoC: sdm845: fix soundwire stream handling" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: stm32: manage rebind issue" from Olivier Moysan <olivier.moysan(a)st.com>: Merge branch 'topic/ro_wordlength' of git://git.kernel.org/.../vkoul/soundwire into asoc-5.7 Merge series "Support built-in Mic on Tegra boards that use WM8903" from Dmitry Osipenko <digetx(a)gmail.com>: ASoC: pxa: Select regmap from AC'97 machines ASoC: pxa: Enable AC'97 bus support for PXA machines Merge series "ASoC: SOF: cleanups and improvements" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt1308-sdw: configure amplifier with set_tdm_slot()" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: add SoundWire support" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: remove rtd->cpu/codec_dai{s}" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge branch 'for-5.6' of https://git.kernel.org/.../broonie/sound into asoc-5.7 Merge series "ASoC: Intel: add SoundWire machine driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'asoc-5.6' into asoc-linus Merge branch 'asoc-5.7' into asoc-next Martin Blumenstingl (3): ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings ASoC: meson: aiu: introduce a struct for platform specific information ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families Nathan Chancellor (1): ASoC: wcd934x: Remove some unnecessary NULL checks Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Oder Chiou (8): ASoC: rt5682: Add the field "is_sdw" of private data ASoC: rt5682: Add the soundwire support ASoC: rt5682: Revise the function name ASoC: rt5682: Revise the DAC1 volume setting ASoC: rt5682: Fine tune the HP performance in soundwire mode ASoC: rt5682: Add a property for DMIC clock rate ASoC: rt5682: Add a property for DMIC delay ASoC: rt5682: Add the descriptions for the DMIC clock rate and delay settings Olivier Moysan (11): ASoC: stm32: sai: manage error when getting reset controller ASoC: stm32: spdifrx: manage error when getting reset controller ASoC: stm32: i2s: manage error when getting reset controller ASoC: stm32: sai: improve error management on probe deferral ASoC: stm32: spdifrx: improve error management on probe deferral ASoC: stm32: i2s: improve error management on probe deferral ASoC: dt-bindings: stm32: convert i2s to json-schema ASoC: dt-bindings: stm32: convert spdfirx to json-schema ASoC: stm32: spdifrx: fix regmap status check ASoC: stm32: spdifrx: manage rebind issue ASoC: stm32: i2s: manage rebind issue Paul Cercueil (6): ASoC: jz4740-i2s: Fix divider written at incorrect offset in register ASoC: jz4740-i2s: Add local dev variable in probe function ASoC: jz4740-i2s: Avoid passing enum as match data ASoC: jz4740-i2s: Add support for the JZ4760 ASoC: jz4740-i2s: Add support for the JZ4770 ASoC: Convert jz4740-i2s doc to YAML Paul Olaru (3): ASoC: SOF: Rename i.MX8 platform to i.MX8X ASoC: SOF: imx8: Add ops for i.MX8QM ASoC: SOF: Add i.MX8QM device descriptor Peter Ujfalusi (5): ASoC: ti: Add udma-pcm platform driver for UDMA ASoC: ti: davinci-mcasp: Add support for platforms using UDMA ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback ALSA: dmaengine_pcm: No need to take runtime reference twice in pcm_pointer Pierre-Louis Bossart (20): ASoC: Intel: sof_pcm512x: add support for SOF platforms with pcm512x ASoC: Intel: BXT: switch pcm512x based boards to sof_pcm512x ASoC: Intel: CHT: add support for pcm512x boards ASoC: soc-dai: add get_sdw_stream() callback ASoC: soc-core: disable route checks for legacy devices ASoC: Intel: skl_nau88l25_ssm4567: disable route checks ASoC: rt5682: fix unmet dependencies ASoC: rt5682-sdw: fix 'defined but not used' pm functions ASoC: Intel: don't use GFP_ATOMIC for machine driver contexts ASoC: rt1308-sdw: add set_tdm_slot() support ASoC: rt1308-sdw: use slot and rx_mask to configure stream ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (3): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms Ranjani Sridharan (11): ASoC: SOF: Do not reset hw_params for streams that ignored suspend ASoC: SOF: pm: Unify suspend/resume routines ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev ASoC: SOF: pm: Introduce DSP power states ASoC: SOF: Move DSP power state transitions to platform-specific ops ASoC: SOF: audio: Add helper to check if only D0i3 streams are active ASoC: SOF: Intel: hda: Amend the DSP state transition diagram ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op ASoC: SOF: Intel: hda: Improve DSP state logging Ravulapati Vishnu vardhan rao (1): ASoC: amd: Adding TDM support in hw_params. Rob Herring (1): ASoC: dt-bindings: google, cros-ec-codec: Fix dtc warnings in example Robin Murphy (2): ASoC: dt-bindings: Make RK3328 codec GPIO explicit ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit Samuel Holland (4): ASoC: sun8i-codec: Remove unused dev from codec struct ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Sathyanarayana Nujella (2): ASoC: Intel: sof_rt5682: Add support for tgl-max98373-rt5682 ASoC: Intel: common: Add mach table for tgl-max98373-rt5682 Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM Shuming Fan (3): ASoC: rt5682: fix the random recording noise of headset ASoC: rt5682: remove noisy debug messages ASoC: rt5682: move DAI clock registry to I2S mode Srinivas Kandagatla (10): ASoC: wcd934x: fix High Accuracy Buck enable ASoC: wcd934x: remove unused headers ASoC: wcd9335: fix address map representation ASoC: qdsp6: q6asm-dai: only enable dais from device tree ASoC: qdsp6: q6routing: remove default routing ASoC: codecs: wsa881x: request gpio direction before setting ASoC: soc-dai: return proper error for get_sdw_stream() ASoC: codecs: wsa881x: remove soundwire stream handling ASoC: qcom: sdm845: handle soundwire stream soundwire: stream: Add read_only_wordlength flag to port properties Stephan Gerhold (1): ASoC: soc-pcm: fix regression in soc_new_pcm() Takashi Iwai (2): ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output ASoC: wm_adsp: Use scnprintf() for the limited buffer output Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp Tzung-Bi Shih (12): ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting drm/mediatek: exit earlier if failed to register audio driver drm/mediatek: support HDMI jack status reporting ASoC: max98357a: move control of SD_MODE to DAPM ASoC: mediatek: mt8183-da7219: add speaker switch ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out ASoC: mediatek: mt8183-da7219: apply some refactors drm/mediatek: fix race condition for HDMI jack status reporting ASoC: dapm: select sleep_state when initializing PINCTRL widget ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out Vinod Koul (9): ALSA: compress: add wma codec profiles ALSA: compress: Add wma decoder params ASoC: qcom: q6asm: pass codec profile to q6asm_open_write ASoC: qcom: q6asm: add support to wma config ASoC: qcom: q6asm-dai: add support to wma decoder ALSA: compress: add alac & ape decoder params ASoC: qcom: q6asm: add support for alac and ape configs ASoC: qcom: q6asm-dai: add support for ALAC and APE decoders ALSA: compress: bump the version Wolfram Sang (1): ASoC: pxa: magician: convert to use i2c_new_client_device() Yong Zhi (4): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support ASoC: max98357a: Add ACPI HID MAX98360A ASoC: intel: sof_da7219_max98373: Add speaker switch Yu-Hsuan Hsu (1): ASoC: cros_ec_codec: Support setting bclk ratio YueHaibing (6): ASoC: Remove unused including <linux/version.h> ASoC: wcd934x: Remove set but not unused variable 'hph_comp_ctrl7' ASoC: rt5682: Make rt5682_clock_config static ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver ASoc: amd: acp3x: Add missing include <linux/io.h> ASoC: wm8974: remove unused variables derek.fang (1): ASoC: rt5682: Enable PLL2 function kbuild test robot (1): ASoC: meson: aiu: fix semicolon.cocci warnings tangbin (2): ASoC: zte: zx-spdif: remove redundant dev_err message ASoC: zte: zx-tdm: remove redundant variables dev .../devicetree/bindings/sound/amlogic,aiu.yaml | 113 ++ .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 + .../bindings/sound/amlogic,gx-sound-card.yaml | 113 ++ .../devicetree/bindings/sound/amlogic,t9015.yaml | 58 + .../bindings/sound/brcm,bcm63xx-audio.txt | 29 + .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 69 + .../devicetree/bindings/sound/cs42l51.txt | 33 - .../bindings/sound/google,cros-ec-codec.txt | 44 - .../bindings/sound/google,cros-ec-codec.yaml | 67 + .../devicetree/bindings/sound/ingenic,aic.yaml | 92 ++ .../bindings/sound/ingenic,jz4740-i2s.txt | 23 - .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 1 + .../bindings/sound/rockchip,rk3328-codec.txt | 7 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 49 - .../devicetree/bindings/sound/rockchip-i2s.yaml | 111 ++ Documentation/devicetree/bindings/sound/rt5682.txt | 18 + .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 - .../devicetree/bindings/sound/st,stm32-i2s.yaml | 87 ++ .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 - .../bindings/sound/st,stm32-spdifrx.yaml | 80 + .../devicetree/bindings/sound/tas2562.txt | 2 +- .../devicetree/bindings/sound/tlv320adcx140.yaml | 82 + Documentation/sound/soc/codec-to-codec.rst | 9 +- MAINTAINERS | 15 +- drivers/gpu/drm/mediatek/mtk_hdmi.c | 54 +- drivers/soundwire/qcom.c | 7 +- drivers/soundwire/stream.c | 16 +- drivers/spi/Kconfig | 2 +- include/dt-bindings/sound/meson-aiu.h | 18 + include/dt-bindings/sound/meson-g12a-toacodec.h | 10 + include/linux/soundwire/sdw.h | 2 + include/sound/compress_driver.h | 40 +- include/sound/hdaudio.h | 2 + include/sound/pcm.h | 14 +- include/sound/rt5682.h | 10 + include/sound/soc-acpi.h | 39 +- include/sound/soc-dai.h | 44 +- include/sound/soc-dapm.h | 6 + include/sound/soc-dpcm.h | 20 +- include/sound/soc.h | 44 +- include/sound/sof/dai-intel.h | 18 +- include/sound/sof/header.h | 11 + include/sound/sof/info.h | 22 +- include/sound/sof/topology.h | 3 +- include/uapi/sound/compress_offload.h | 2 +- include/uapi/sound/compress_params.h | 37 +- include/uapi/sound/sof/abi.h | 2 +- sound/arm/pxa2xx-pcm-lib.c | 8 +- sound/core/compress_offload.c | 42 + sound/core/pcm_dmaengine.c | 6 +- sound/core/pcm_misc.c | 18 +- sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 376 +++++ sound/soc/amd/raven/acp3x-i2s.c | 44 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/amd/raven/pci-acp3x.c | 7 +- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 4 + sound/soc/bcm/bcm63xx-i2s-whistler.c | 317 ++++ sound/soc/bcm/bcm63xx-i2s.h | 90 ++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 485 ++++++ sound/soc/bcm/cygnus-pcm.c | 22 +- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/Kconfig | 627 ++++---- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cros_ec_codec.c | 25 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/cs47l35.c | 6 +- sound/soc/codecs/cs47l85.c | 6 +- sound/soc/codecs/cs47l90.c | 6 +- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/codecs/max98357a.c | 37 +- sound/soc/codecs/mt6660.c | 81 +- sound/soc/codecs/rk3328_codec.c | 31 +- sound/soc/codecs/rl6231.c | 1 + sound/soc/codecs/rl6231.h | 2 +- sound/soc/codecs/rt1015.c | 10 +- sound/soc/codecs/rt1308-sdw.c | 38 +- sound/soc/codecs/rt1308-sdw.h | 2 + sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 333 ++++ sound/soc/codecs/rt5682-sdw.h | 20 + sound/soc/codecs/rt5682.c | 1298 ++++++++++++++-- sound/soc/codecs/rt5682.h | 100 +- sound/soc/codecs/tas2562.c | 121 +- sound/soc/codecs/tas2562.h | 12 +- sound/soc/codecs/tlv320adcx140.c | 920 +++++++++++ sound/soc/codecs/tlv320adcx140.h | 131 ++ sound/soc/codecs/wcd9335.c | 18 +- sound/soc/codecs/wcd9335.h | 7 +- sound/soc/codecs/wcd934x.c | 37 +- sound/soc/codecs/wm0010.c | 2 +- sound/soc/codecs/wm5110.c | 6 +- sound/soc/codecs/wm8974.c | 8 - sound/soc/codecs/wm_adsp.c | 14 +- sound/soc/codecs/wsa881x.c | 46 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/fsl_asrc_dma.c | 10 +- sound/soc/fsl/fsl_spdif.c | 10 +- sound/soc/fsl/fsl_ssi.c | 8 +- sound/soc/fsl/imx-audmix.c | 8 +- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +- sound/soc/generic/simple-card-utils.c | 60 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/atom/sst/sst_pci.c | 2 +- sound/soc/intel/boards/Kconfig | 57 +- sound/soc/intel/boards/Makefile | 12 +- sound/soc/intel/boards/bdw-rt5650.c | 15 +- sound/soc/intel/boards/bdw-rt5677.c | 15 +- sound/soc/intel/boards/broadwell.c | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 10 +- sound/soc/intel/boards/bxt_rt298.c | 10 +- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +- sound/soc/intel/boards/bytcht_da7213.c | 10 +- sound/soc/intel/boards/bytcht_es8316.c | 8 +- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 8 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 8 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 16 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 13 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 13 +- sound/soc/intel/boards/haswell.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 14 +- sound/soc/intel/boards/kbl_rt5660.c | 6 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 12 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 12 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 4 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 27 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 +- sound/soc/intel/boards/skl_rt286.c | 8 +- sound/soc/intel/boards/sof_da7219_max98373.c | 83 +- sound/soc/intel/boards/sof_maxim_common.c | 80 + sound/soc/intel/boards/sof_maxim_common.h | 24 + sound/soc/intel/boards/sof_pcm512x.c | 448 ++++++ sound/soc/intel/boards/sof_rt5682.c | 137 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 ++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 ++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 ++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 ++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 ++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 7 + sound/soc/intel/common/soc-acpi-intel-cml-match.c | 111 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 103 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 34 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 92 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 +- sound/soc/intel/skylake/bxt-sst.c | 3 - sound/soc/intel/skylake/cnl-sst.c | 35 +- sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-pcm.c | 20 +- sound/soc/intel/skylake/skl-sst-dsp.h | 2 + sound/soc/intel/skylake/skl.c | 33 +- sound/soc/jz4740/jz4740-i2s.c | 78 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 117 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/Kconfig | 41 + sound/soc/meson/Makefile | 19 + sound/soc/meson/aiu-acodec-ctrl.c | 203 +++ sound/soc/meson/aiu-codec-ctrl.c | 151 ++ sound/soc/meson/aiu-encoder-i2s.c | 365 +++++ sound/soc/meson/aiu-encoder-spdif.c | 209 +++ sound/soc/meson/aiu-fifo-i2s.c | 153 ++ sound/soc/meson/aiu-fifo-spdif.c | 186 +++ sound/soc/meson/aiu-fifo.c | 223 +++ sound/soc/meson/aiu-fifo.h | 50 + sound/soc/meson/aiu.c | 388 +++++ sound/soc/meson/aiu.h | 89 ++ sound/soc/meson/axg-card.c | 414 +---- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/g12a-toacodec.c | 252 +++ sound/soc/meson/g12a-tohdmitx.c | 219 +-- sound/soc/meson/gx-card.c | 141 ++ sound/soc/meson/meson-card-utils.c | 385 +++++ sound/soc/meson/meson-card.h | 55 + sound/soc/meson/meson-codec-glue.c | 149 ++ sound/soc/meson/meson-codec-glue.h | 32 + sound/soc/meson/t9015.c | 333 ++++ sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/Kconfig | 22 + sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 14 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +- sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/apq8016_sbc.c | 9 +- sound/soc/qcom/apq8096.c | 6 +- sound/soc/qcom/lpass-platform.c | 4 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 173 ++- sound/soc/qcom/qdsp6/q6asm.c | 243 ++- sound/soc/qcom/qdsp6/q6asm.h | 51 +- sound/soc/qcom/qdsp6/q6routing.c | 21 +- sound/soc/qcom/sdm845.c | 105 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +- sound/soc/rockchip/rockchip_max98090.c | 6 +- sound/soc/rockchip/rockchip_rt5645.c | 6 +- sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/arndale.c | 10 +- sound/soc/samsung/bells.c | 16 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 16 +- sound/soc/samsung/lowland.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 10 +- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 4 +- sound/soc/samsung/smdk_wm8994pcm.c | 6 +- sound/soc/samsung/snow.c | 8 +- sound/soc/samsung/spdif.c | 8 +- sound/soc/samsung/speyside.c | 10 +- sound/soc/samsung/tm2_wm5110.c | 19 +- sound/soc/samsung/tobermory.c | 10 +- sound/soc/sh/dma-sh7760.c | 16 +- sound/soc/sh/fsi.c | 5 +- sound/soc/sh/migor.c | 6 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 290 ++-- sound/soc/soc-dai.c | 18 +- sound/soc/soc-dapm.c | 220 +-- sound/soc/soc-generic-dmaengine-pcm.c | 26 +- sound/soc/soc-pcm.c | 1627 +++++++++----------- sound/soc/soc-topology.c | 24 +- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 ++ sound/soc/sof/compress.h | 31 + sound/soc/sof/core.c | 10 +- sound/soc/sof/debug.c | 226 +++ sound/soc/sof/imx/imx8.c | 57 +- sound/soc/sof/intel/Kconfig | 20 +- sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 51 +- sound/soc/sof/intel/hda-codec.c | 11 +- sound/soc/sof/intel/hda-compress.c | 114 ++ sound/soc/sof/intel/hda-ctrl.c | 40 +- sound/soc/sof/intel/hda-dai.c | 130 +- sound/soc/sof/intel/hda-dsp.c | 331 +++- sound/soc/sof/intel/hda-ipc.c | 24 +- sound/soc/sof/intel/hda-loader.c | 40 +- sound/soc/sof/intel/hda-pcm.c | 8 +- sound/soc/sof/intel/hda-stream.c | 27 +- sound/soc/sof/intel/hda.c | 433 +++++- sound/soc/sof/intel/hda.h | 120 +- sound/soc/sof/ipc.c | 41 +- sound/soc/sof/loader.c | 6 - sound/soc/sof/ops.h | 59 +- sound/soc/sof/pcm.c | 19 +- sound/soc/sof/pm.c | 176 +-- sound/soc/sof/probe.c | 290 ++++ sound/soc/sof/probe.h | 85 + sound/soc/sof/sof-audio.c | 59 +- sound/soc/sof/sof-audio.h | 6 +- sound/soc/sof/sof-of-dev.c | 10 + sound/soc/sof/sof-priv.h | 71 +- sound/soc/sof/topology.c | 25 +- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +- sound/soc/stm/stm32_i2s.c | 75 +- sound/soc/stm/stm32_sai.c | 26 +- sound/soc/stm/stm32_sai_sub.c | 13 +- sound/soc/stm/stm32_spdifrx.c | 89 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 3 - sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 24 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/Kconfig | 8 +- sound/soc/ti/Makefile | 2 + sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-mcasp.c | 13 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/ti/udma-pcm.c | 43 + sound/soc/ti/udma-pcm.h | 18 + sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 6 +- sound/soc/ux500/ux500_pcm.c | 8 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 1 - sound/soc/zte/zx-tdm.c | 3 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 379 files changed, 17570 insertions(+), 3797 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,aiu.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/ingenic,aic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 include/dt-bindings/sound/meson-aiu.h create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c create mode 100644 sound/soc/bcm/bcm63xx-i2s-whistler.c create mode 100644 sound/soc/bcm/bcm63xx-i2s.h create mode 100644 sound/soc/bcm/bcm63xx-pcm-whistler.c create mode 100644 sound/soc/codecs/rt5682-sdw.c create mode 100644 sound/soc/codecs/rt5682-sdw.h create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h create mode 100644 sound/soc/intel/boards/sof_maxim_common.c create mode 100644 sound/soc/intel/boards/sof_maxim_common.h create mode 100644 sound/soc/intel/boards/sof_pcm512x.c create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c create mode 100644 sound/soc/meson/aiu-acodec-ctrl.c create mode 100644 sound/soc/meson/aiu-codec-ctrl.c create mode 100644 sound/soc/meson/aiu-encoder-i2s.c create mode 100644 sound/soc/meson/aiu-encoder-spdif.c create mode 100644 sound/soc/meson/aiu-fifo-i2s.c create mode 100644 sound/soc/meson/aiu-fifo-spdif.c create mode 100644 sound/soc/meson/aiu-fifo.c create mode 100644 sound/soc/meson/aiu-fifo.h create mode 100644 sound/soc/meson/aiu.c create mode 100644 sound/soc/meson/aiu.h create mode 100644 sound/soc/meson/g12a-toacodec.c create mode 100644 sound/soc/meson/gx-card.c create mode 100644 sound/soc/meson/meson-card-utils.c create mode 100644 sound/soc/meson/meson-card.h create mode 100644 sound/soc/meson/meson-codec-glue.c create mode 100644 sound/soc/meson/meson-codec-glue.h create mode 100644 sound/soc/meson/t9015.c create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h create mode 100644 sound/soc/ti/udma-pcm.c create mode 100644 sound/soc/ti/udma-pcm.h

2 1

[PATCH 0/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

Hi, Mark I resend the patch and add recipient Vinod as your comment. Thank you. I'd like to send a patch to fix a problem using soc-compress with DPCM. soc_compr_open_fe() and soc_compr_free_fe() call snd_soc_runtime_activate() and snd_soc_runtime_deactivate() but don't lock card->pcm_mutex. It can cause lockdep error, because snd_soc_runtime_activate/deactivate() checks whether pcm_mutex is held using lockdep_assert_held(). I'd like to send a patch which adds mutex_lock/unlock() before and after of the snd_soc_runtime_activate/deactivate() call. If there is anything should be changed in my patch or email, please let me know it. I'll really appreciate it. Lee Gyeongtaek Lee (1): ASoC: soc-compress: lock pcm_mutex to resolve lockdep error sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 1/1] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by 이경택 30 Mar '20

30 Mar '20

snd_soc_runtime_activate() and snd_soc_runtime_deactivate() require pcm_mutex lock. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..42d416ac7e9b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -207,7 +207,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -285,7 +287,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) else stream = SNDRV_PCM_STREAM_CAPTURE; + mutex_lock_nested(&fe->pcm_mutex, fe->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.21.0

2 2

[RFC PATCH v1 13/50] Avoid some useless msecs/jiffies conversions
by George Spelvin 30 Mar '20

30 Mar '20

Rather than generating a random number of milliseconds in a constant range and then converting to jiffies, convert the range to jiffies (evaluated at compile time) and choose a random number of jiffies in that range. Likewise, "msecs_to_jiffies(seconds * 1000)" is more conveniently written "seconds * HZ". Signed-off-by: George Spelvin <lkml(a)sdf.org> Cc: Hannes Reinecke <hare(a)suse.de> Cc: linux-scsi(a)vger.kernel.org Cc: Marek Lindner <mareklindner(a)neomailbox.ch> Cc: Simon Wunderlich <sw(a)simonwunderlich.de> Cc: Antonio Quartulli <a(a)unstable.cc> Cc: Sven Eckelmann <sven(a)narfation.org> Cc: b.a.t.m.a.n(a)lists.open-mesh.org Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-wireless(a)vger.kernel.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- drivers/scsi/fcoe/fcoe_ctlr.c | 10 +++++----- net/batman-adv/bat_iv_ogm.c | 2 +- net/batman-adv/bat_v_ogm.c | 8 ++++---- net/bluetooth/hci_request.c | 2 +- net/wireless/scan.c | 2 +- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 7 files changed, 14 insertions(+), 14 deletions(-) diff --git a/drivers/scsi/fcoe/fcoe_ctlr.c b/drivers/scsi/fcoe/fcoe_ctlr.c index 1791a393795da..9c530f8827518 100644 --- a/drivers/scsi/fcoe/fcoe_ctlr.c +++ b/drivers/scsi/fcoe/fcoe_ctlr.c @@ -2238,10 +2238,10 @@ static void fcoe_ctlr_vn_restart(struct fcoe_ctlr *fip) if (fip->probe_tries < FIP_VN_RLIM_COUNT) { fip->probe_tries++; - wait = prandom_u32() % FIP_VN_PROBE_WAIT; + wait = prandom_u32_max(msecs_to_jiffies(FIP_VN_PROBE_WAIT)); } else - wait = FIP_VN_RLIM_INT; - mod_timer(&fip->timer, jiffies + msecs_to_jiffies(wait)); + wait = msecs_to_jiffies(FIP_VN_RLIM_INT); + mod_timer(&fip->timer, jiffies + wait); fcoe_ctlr_set_state(fip, FIP_ST_VNMP_START); } @@ -3132,8 +3132,8 @@ static void fcoe_ctlr_vn_timeout(struct fcoe_ctlr *fip) fcoe_ctlr_vn_send(fip, FIP_SC_VN_BEACON, fcoe_all_vn2vn, 0); fip->port_ka_time = jiffies + - msecs_to_jiffies(FIP_VN_BEACON_INT + - (prandom_u32() % FIP_VN_BEACON_FUZZ)); + msecs_to_jiffies(FIP_VN_BEACON_INT) + + prandom_u32_max(msecs_to_jiffies(FIP_VN_BEACON_FUZZ)); } if (time_before(fip->port_ka_time, next_time)) next_time = fip->port_ka_time; diff --git a/net/batman-adv/bat_iv_ogm.c b/net/batman-adv/bat_iv_ogm.c index 46c5cd9f8019e..9efd96e91d53e 100644 --- a/net/batman-adv/bat_iv_ogm.c +++ b/net/batman-adv/bat_iv_ogm.c @@ -288,7 +288,7 @@ batadv_iv_ogm_emit_send_time(const struct batadv_priv *bat_priv) /* when do we schedule a ogm packet to be sent */ static unsigned long batadv_iv_ogm_fwd_send_time(void) { - return jiffies + msecs_to_jiffies(prandom_u32() % (BATADV_JITTER / 2)); + return jiffies + prandom_u32_max(msecs_to_jiffies(BATADV_JITTER / 2)); } /* apply hop penalty for a normal link */ diff --git a/net/batman-adv/bat_v_ogm.c b/net/batman-adv/bat_v_ogm.c index 411ce5fc6492f..61fa742ff5501 100644 --- a/net/batman-adv/bat_v_ogm.c +++ b/net/batman-adv/bat_v_ogm.c @@ -85,12 +85,12 @@ struct batadv_orig_node *batadv_v_ogm_orig_get(struct batadv_priv *bat_priv, */ static void batadv_v_ogm_start_queue_timer(struct batadv_hard_iface *hard_iface) { - unsigned int msecs = BATADV_MAX_AGGREGATION_MS * 1000; + unsigned int delay = msecs_to_jiffies(BATADV_MAX_AGGREGATION_MS); - /* msecs * [0.9, 1.1] */ - msecs += prandom_u32() % (msecs / 5) - (msecs / 10); + /* delay * [0.9, 1.1] */ + delay += prandom_u32_max(delay / 5) - (delay / 10); queue_delayed_work(batadv_event_workqueue, &hard_iface->bat_v.aggr_wq, - msecs_to_jiffies(msecs / 1000)); + delay); } /** diff --git a/net/bluetooth/hci_request.c b/net/bluetooth/hci_request.c index 2a1b64dbf76e4..8b46e23b4abe7 100644 --- a/net/bluetooth/hci_request.c +++ b/net/bluetooth/hci_request.c @@ -1505,7 +1505,7 @@ int hci_get_random_address(struct hci_dev *hdev, bool require_privacy, bacpy(rand_addr, &hdev->rpa); - to = msecs_to_jiffies(hdev->rpa_timeout * 1000); + to = hdev->rpa_timeout * HZ; if (adv_instance) queue_delayed_work(hdev->workqueue, &adv_instance->rpa_expired_cb, to); diff --git a/net/wireless/scan.c b/net/wireless/scan.c index aef240fdf8df6..b6856cbb68d3b 100644 --- a/net/wireless/scan.c +++ b/net/wireless/scan.c @@ -700,7 +700,7 @@ void cfg80211_bss_age(struct cfg80211_registered_device *rdev, unsigned long age_secs) { struct cfg80211_internal_bss *bss; - unsigned long age_jiffies = msecs_to_jiffies(age_secs * MSEC_PER_SEC); + unsigned long age_jiffies = age_secs * HZ; spin_lock_bh(&rdev->bss_lock); list_for_each_entry(bss, &rdev->bss_list, list) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2236b5e0c1f25..8a2bf333200c1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1839,7 +1839,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time *= HZ; } } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index df40d38f6e290..1ea763f9f956d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1937,7 +1937,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, long t = runtime->period_size * 2 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout *= HZ; } tout = schedule_timeout(tout); -- 2.26.0

3 9

[PATCH 1/1] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. To fix this problem, SND_SOC_DPCM_STATE_PAUSED should be allowed also for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

1 0

[PATCH 0/1] ASoC: dpcm: allow started or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

Hi, When compress offload is used with DPCM, error can be occurred on start or stop after pause. soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. Any feedback will be greatly appreciated. Gyeongtaek Lee (1): ASoC: dpcm: allow start or stop during pause for backend sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

1 0

[PATCH 0/4] ASoC: remove rtd->cpu/codec_dai{s} v2
by Kuninori Morimoto 30 Mar '20

30 Mar '20

Hi Mark These are rebased "remove rtd->cpu/codec_dai" patches. Kuninori Morimoto (4): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 26 +++++++++-------- sound/soc/soc-compress.c | 36 +++++++++++------------ sound/soc/soc-core.c | 42 ++++++++++----------------- sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++-- sound/soc/soc-pcm.c | 30 +++++++++---------- 6 files changed, 68 insertions(+), 76 deletions(-) -- 2.17.1

1 4

[PATCH v2] fix some output pcms being namehint-ed as input pcms
by sylvain.bertrand＠gmail.com 29 Mar '20

29 Mar '20

This is additional cleanup following: commit d434638a683c01bd6decf8f41863044055237a50 namehint: remember the direction from the upper levels signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- based on arecord -L output --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

[PATCH] ASoC: bcm2835-i2s: substream alignment now independent in hwparams
by Matt Flax 29 Mar '20

29 Mar '20

Substream sample alignment was being set in hwparams for both substreams at the same time. This became a problem when the Audio Injector isolated sound card needed to offset sample alignment for high sample rates. The latency difference between playback and capture occurs because of the digital isolation chip propagation time, particularly when the codec is master and the DAC return is twice delayed. This patch sets sample alignment registers based on the substream direction in hwparams. This gives the machine driver more control over sample alignment in the bcm2835 i2s driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 36 +++++++++++++++++++----------------- 1 file changed, 19 insertions(+), 17 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e6a12e271b07..9db542699a13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -493,11 +493,6 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, - rx_mask, slot_width, data_delay, odd_slot_offset); - bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, - tx_mask, slot_width, data_delay, odd_slot_offset); - /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable @@ -508,19 +503,26 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, "Unstable slave config detected, L/R may be swapped"); /* - * Set format for both streams. - * We cannot set another frame length - * (and therefore word length) anyway, - * so the format will be the same. + * Set format on a per stream basis. + * The alignment format can be different depending on direction. */ - regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, - format - | BCM2835_I2S_CH1_POS(rx_ch1_pos) - | BCM2835_I2S_CH2_POS(rx_ch2_pos)); - regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, - format - | BCM2835_I2S_CH1_POS(tx_ch1_pos) - | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, + rx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, + format + | BCM2835_I2S_CH1_POS(rx_ch1_pos) + | BCM2835_I2S_CH2_POS(rx_ch2_pos)); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, + tx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, + format + | BCM2835_I2S_CH1_POS(tx_ch1_pos) + | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + } /* Setup the I2S mode */ -- 2.20.1

3 8

[PATCH v2] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 29 Mar '20

29 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

2 1

about alc880 no sound and eapd coef
by 尤晓杰 29 Mar '20

29 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd? the attachment is the schematic of sound part.in summary: the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out, no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. I what to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13

2 1

Port HDA Analyzer to python 3
by andythe_great 29 Mar '20

29 Mar '20

Hello, Is it possible to port HDA Analyzer to python 3? https://www.alsa-project.org/main/index.php/HDA_Analyzer Because openSUSE is retiring python 2. I tried using python-modernize https://python-modernize.readthedocs.io/en/latest/ but many could not get it to run. Thanks. Andy

2 1

[PATCH] ALSA: hda: default enable CA0132 DSP support
by Rouven Czerwinski 29 Mar '20

29 Mar '20

From: Rouven Czerwinski <rouven(a)czerwinskis.de> If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski <rouven(a)czerwinskis.de> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help base-commit: e595dd94515ed6bc5ba38fce0f9598db8c0ee9a9 -- 2.25.1

2 1

[alsa-devel] [PATCH 0/3] ALSA: bebob: misc improvements for starting/stopping packet streaming
by Takashi Sakamoto 27 Mar '20

27 Mar '20

Hi, This patchset improves model-specific issues in a point of protocol version supported by the model. The changes are relevant to packet streaming, thus it can brings some regressions. For testers, I prepare for a remote branch to backport the latest codes to Linux v4.17 or later. I'm happy to receive any report. https://github.com/takaswie/snd-firewire-improve/tree/topic/bebob-connectio… Takashi Sakamoto (3): ALSA: bebob: expand sleep just after breaking connections for protocol version 1 ALSA: bebob: expand delay of start for IR context just for version 3 firmware ALSA: bebob: link the order of establishing connections and Syt-match clock mode sound/firewire/bebob/bebob_stream.c | 76 ++++++++++++++--------------- 1 file changed, 38 insertions(+), 38 deletions(-) -- 2.20.1

3 7

[PATCH] ASoC: topology: Add missing memory checks
by Amadeusz Sławiński 27 Mar '20

27 Mar '20

kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 65 +++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..0bec3ff782c1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1857,9 +1867,17 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (ret != 0) { dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret); snd_soc_unregister_dai(dai); - return ret; + goto err; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,16 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + + return ret; } /* load physical DAI elements */ -- 2.17.1

2 2

[PATCH 0/4] ASoC: Intel: add SoundWire machine driver
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

To handle multiple hardware combinations, this patchset suggests a single machine driver which will create and initialize dailinks dynamically. This allows us to support new configurations easily, as shown with the TigerLake rt5682 example. Each configuration updates the card component string, and UCM can test for the presence of components to configure them as needed. Since we use a single the machine driver name, all previous ACPI tables need to be updated. That should have no impact since the machine drivers listed at the time were not upstreamed and are no longer maintained. Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Pierre-Louis Bossart (1): ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (1): ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 8 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + .../intel/common/soc-acpi-intel-cml-match.c | 24 +- .../intel/common/soc-acpi-intel-icl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 30 +- 14 files changed, 1896 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c -- 2.20.1

2 11

[PATCH] dt-bindings: sound: tlv320adcx140: Remove undocumented property
by Dan Murphy 27 Mar '20

27 Mar '20

Remove undocumented and unneeded ti,use-internal-reg from the example as it was an artifact from initial development. The code does not query for this property and as the document indicates if areg-supply is undefined then the internal regulator is used. Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt bindings") CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 1433ff62b14f..ab2268c0ee67 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -76,7 +76,6 @@ examples: codec: codec@4c { compatible = "ti,tlv320adc5140"; reg = <0x4c>; - ti,use-internal-areg; ti,mic-bias-source = <6>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; -- 2.25.1

2 1

[PATCH] ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
by Colin King 27 Mar '20

27 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of statements that are not indented correctly, add in the missing tab and break the lines to address a checkpatch warning. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e..e362f0bc9e46 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); -- 2.25.1

2 1

[PATCH] additional cleanup for "arecord -L show entries which are not capture pcms"
by sylvain.bertrand＠gmail.com 27 Mar '20

27 Mar '20

I did post to github, my bad. On Fri, Mar 27, 2020 at 04:01:57AM -0700, Jaroslav Kysela wrote: > Closed #39 via d434638a683c01bd6decf8f41863044055237a50. Tested with the mic from my usb webcam, and it actually needs a little bit more tidying. -- Sylvain --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

route and chmap
by Jonas Holmberg 27 Mar '20

27 Mar '20

Hi, I use the route plugin to copy mono to both FL and FR (front left and front right stereo) of the slave during playback. When get_chmap() is called in the route plugin it will call get_chmap() of the slave, which returns 2 channels with FL and FR positions, and then route itself returns 1 channel with FL position. But the application that called snd_pcm_get_chmap() on the playback device expected to get position MONO instead of FL since the number of channels is 1. Is the application wrong in not accepting the chmap or should the route plugin have changed position from FL to MONO? Is it possible to add something to the config (asound.conf) to make FL become MONO in the reply of get_chmap() or solve the problem in any other way? BR /Jonas

2 2

[PATCH 00/36] ASoC: remove rtd->cpu/codec_dai{s}
by Kuninori Morimoto 27 Mar '20

27 Mar '20

Hi Mark Now, CPU/Codec DAI(s) were replaced by rtd->dais. Thus, We don't need rtd->cpu/codec_dai{s} anymore. This pathset replaces it by new macro. Kuninori Morimoto (36): ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 30 +++++++------ sound/arm/pxa2xx-pcm-lib.c | 8 ++-- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 16 +++---- sound/soc/bcm/cygnus-pcm.c | 22 +++++----- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +-- sound/soc/codecs/cs47l35.c | 6 +-- sound/soc/codecs/cs47l85.c | 6 +-- sound/soc/codecs/cs47l90.c | 6 +-- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/wm5110.c | 6 +-- sound/soc/codecs/wm_adsp.c | 10 ++--- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 ++--- sound/soc/fsl/fsl_asrc_dma.c | 6 +-- sound/soc/fsl/fsl_spdif.c | 10 ++--- sound/soc/fsl/fsl_ssi.c | 8 ++-- sound/soc/fsl/imx-audmix.c | 8 ++-- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 ++--- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +-- sound/soc/generic/simple-card-utils.c | 12 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +-- sound/soc/intel/boards/bdw-rt5650.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 6 +-- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/bxt_rt298.c | 8 ++-- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 ++--- sound/soc/intel/boards/bytcht_da7213.c | 8 ++-- sound/soc/intel/boards/bytcht_es8316.c | 8 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +-- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 ++-- sound/soc/intel/boards/cml_rt1011_rt5682.c | 6 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 ++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +-- sound/soc/intel/boards/kbl_rt5660.c | 6 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +++---- sound/soc/intel/boards/skl_rt286.c | 8 ++-- sound/soc/intel/boards/sof_da7219_max98373.c | 8 ++-- sound/soc/intel/boards/sof_pcm512x.c | 8 ++-- sound/soc/intel/boards/sof_rt5682.c | 6 +-- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 ++++++------ sound/soc/intel/skylake/skl-pcm.c | 10 ++--- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 ++--- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 6 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 4 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-card.c | 8 ++-- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 8 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 6 +-- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 22 +++++----- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +++---- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/rockchip/rockchip_rt5645.c | 6 +-- sound/soc/samsung/arndale.c | 6 +-- sound/soc/samsung/bells.c | 16 +++---- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 14 +++---- sound/soc/samsung/lowland.c | 4 +- sound/soc/samsung/neo1973_wm8753.c | 10 ++--- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +-- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 4 +- sound/soc/samsung/spdif.c | 8 ++-- sound/soc/samsung/speyside.c | 8 ++-- sound/soc/samsung/tm2_wm5110.c | 16 +++---- sound/soc/samsung/tobermory.c | 8 ++-- sound/soc/sh/dma-sh7760.c | 16 +++---- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 6 +-- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 36 ++++++++-------- sound/soc/soc-core.c | 42 +++++++------------ sound/soc/soc-dapm.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +-- sound/soc/soc-pcm.c | 30 ++++++------- sound/soc/sof/intel/hda-dai.c | 6 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +++--- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 6 +-- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +-- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +++++----- sound/soc/uniphier/aio-dma.c | 6 +-- sound/soc/ux500/mop500_ab8500.c | 6 +-- sound/soc/ux500/ux500_pcm.c | 8 ++-- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 191 files changed, 573 insertions(+), 577 deletions(-) -- 2.17.1

3 72

[PATCH v2 00/11] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series will not be in 5.7, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change in significant ways. Changes since v2: Corrected error in ACPI table (thanks Amadeusz) Added patch 11 to add reset cycle required on some SoundWire platforms Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (8): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-ctrl.c | 25 +- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 400 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 10 files changed, 750 insertions(+), 64 deletions(-) -- 2.20.1

2 14

Applied "ASoC: SOF: Intel: add SoundWire configuration interface" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: add SoundWire configuration interface has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:18 -0500 Subject: [PATCH] ASoC: SOF: Intel: add SoundWire configuration interface Now that the SoundWire core supports the multi-step initialization, call the relevant APIs. The actual hardware enablement can be done in two places, ideally we'd want to startup the SoundWire IP as soon as possible (while still taking power rail dependencies into account) However when suspend/resume is implemented, the DSP device will be resumed first, and only when the DSP firmware is downloaded/booted would the SoundWire child devices be resumed, so there are only marginal benefits in starting the IP earlier for the first probe. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 13 ++++ sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 44 ++++++++++++ 3 files changed, 177 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 0633b76dab49..2ae94ea53122 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee1edb53840c..c1fe94800da1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,7 +18,9 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <linux/acpi.h> #include <linux/module.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> @@ -34,6 +36,98 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + acpi_handle handle; + void *sdw; + + handle = ACPI_HANDLE(sdev->dev); + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} +#endif + /* * Debug */ @@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); + return ret; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + /* + * probe/allocated SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + } + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2a57fc9f3e58..928a3432e9e6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,8 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/compress_driver.h> #include <sound/hda_codec.h> #include <sound/hdaudio_ext.h> @@ -436,6 +438,12 @@ struct sof_intel_hda_dev { /* delayed work to enter D0I3 opportunistically */ struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; -- 2.20.1

1 0

Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: IPC: dai-intel: move ALH declarations in header file has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:19 -0500 Subject: [PATCH] ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ALH was inserted in the wrong place during integration, add after DMIC to mirror the file used by SOF firmware. No functional change, just text move in the same file to better track changes, if any. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/dai-intel.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5f1ef5565be6..04e48227f542 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params { uint32_t link_dma_ch; } __packed; +/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ +struct sof_ipc_dai_alh_params { + struct sof_ipc_hdr hdr; + uint32_t stream_id; + + /* reserved for future use */ + uint32_t reserved[15]; +} __packed; + /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ /* This struct is defined per 2ch PDM controller available in the platform. @@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params { struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; } __packed; -/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ -struct sof_ipc_dai_alh_params { - struct sof_ipc_hdr hdr; - uint32_t stream_id; - - /* reserved for future use */ - uint32_t reserved[15]; -} __packed; - #endif -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:20 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks These callbacks are invoked when a matching hw_params/hw_free() DAI operation takes place, and will result in IPC operations with the SOF firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1fe94800da1..1f93124a63cd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -38,6 +39,74 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); @@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.irq = sdev->ipc_irq; res.handle = hdev->info.handle; res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; /* * ops and arg fields are not populated for now, -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:22 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend Doing this avoid conflicts and errors reported on the bus. The interrupts are only re-enabled on resume after the firmware is downloaded, so the behavior is not fully symmetric Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328..1aff90042694 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001 From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:23 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers We have a single irq handler for SOF interrupts. We can further merge SoundWire ones to completely remove MSI interrupts handling issues leading to timeouts. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index dea3c385b664..ee4f1ceca883 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev) return 0; } + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + #endif /* @@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 928a3432e9e6..fc104c5ba006 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -232,6 +232,8 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { } +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} #endif /* common dai driver */ -- 2.20.1

1 0