September 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm9712: fix unused variable warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

The 'ret' variable is now only used in an #ifdef, and causes a warning if it is declared outside of that block: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable] Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..e873baa9e778 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; } else { #ifdef CONFIG_SND_SOC_AC97_BUS + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Add r8a7744 support
by Biju Das 28 Sep '18

28 Sep '18

Document RZ/G1N (R8A7744) SoC bindings. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> Reviewed-by: Chris Paterson <Chris.Paterson2(a)renesas.com> --- This patch is tested against linux-next next-20180927. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f4688c5..d92b705 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -343,6 +343,7 @@ Required properties: "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) Examples with soctypes are: - "renesas,rcar_sound-r8a7743" (RZ/G1M) + - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - "renesas,rcar_sound-r8a7778" (R-Car M1A) -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include" to the asoc tree
by Mark Brown 28 Sep '18

28 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include 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 06da26e5ce157bf2af09ef4a9f5dc30af4599fa0 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 26 Sep 2018 15:33:03 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include Remove duplicated includes linux/of_platform.h and linux/wait.h Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 2b2c7233bb5f..e1cfa846a1dc 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -11,7 +11,6 @@ #include <linux/spinlock.h> #include <linux/kref.h> #include <linux/of.h> -#include <linux/of_platform.h> #include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index ca1be7305524..cdfc8ab6cfc0 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -10,7 +10,6 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/jiffies.h> -#include <linux/wait.h> #include <linux/soc/qcom/apr.h> #include "q6core.h" #include "q6dsp-errno.h" -- 2.19.0

1 0

Re: [alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by AC30 CTLin0 28 Sep '18

28 Sep '18

OK, I will resend a new patch. However, from my last patch file, I can see DT file where is located at "Documentation/devicetree/bindings/sound/nau8822.txt". And don't you see any DT file? On 9/26/2018 1:52 AM, Mark Brown wrote: > On Tue, Sep 25, 2018 at 09:56:13AM -0700, Mark Brown wrote: >> On Tue, Sep 25, 2018 at 10:50:19AM +0800, AC30 CTLin0 wrote: >>> May I learn the status of patch review due to our customer ask us about >>> upstream status? >> I don't know, I don't seem to have a copy of this patch in my review >> queue. Please resend: > Found a copy in the list archives, but still no idea where it went in my > review queue. My best guess is that it got eaten by spam filtering > somewhere but I'm not 100% sure. Sorry about that. :( The driver > looks pretty clean, a few small points: > > - The Mute Switch controls should just be called Switch so that > applications know how to handle them. > - PGA Boost +20dB should just be "PGA Boost Volume" with the +20dB > specified by having it be a SOC_DOUBLE_R_TLV(). > - The switch statements in hw_params() should have default: > cases that return errors so that if an incorrect parameter makes it > through we detect it. > - There's a DT compatible string but no DT binding file is supplied, > should just be a very simple one as there's no parameters. > > These are all small, hopefully should be quick to update. =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

2 1

[alsa-devel] [PATCH linux-next 08/10] arm64: r8a77965: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index fd00010a5ed7..6c0d5053a81c 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1405,7 +1405,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ /* placeholder */ rcar_sound,dvc { -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 07/10] arm64: r8a774a1: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi index c9aad3e3b1b0..c3e0f2151176 100644 --- a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi @@ -1120,7 +1120,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 06/10] arm64: r8a7796: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index e9b05d997c4e..827a704154cd 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -1670,7 +1670,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 05/10] arm64: r8a7795: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7795.dtsi b/arch/arm64/boot/dts/renesas/r8a7795.dtsi index 0a66bda7124f..32d191102a81 100644 --- a/arch/arm64/boot/dts/renesas/r8a7795.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7795.dtsi @@ -1796,7 +1796,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH 0/5] MAX98088/9 of features
by Marco Felsch 26 Sep '18

26 Sep '18

Hi, I picked up the max98088 patches from Andreas series [1] and ported them to the new kernel. Regards, Marco [1] http://lists.infradead.org/pipermail/linux-arm-kernel/ \ 2015-February/324440.html Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable .../bindings/sound/maxim,max98088.txt | 20 ++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98088.c | 39 +++++++++++++++++++ 3 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt -- 2.19.0

2 11

Re: [alsa-devel] linux-next: Tree for Sep 25 (sound/soc/codecs/wm8731)
by Randy Dunlap 25 Sep '18

25 Sep '18

On 9/24/18 10:39 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20180924: > on i386: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] and then: ld: sound/soc/codecs/wm8731.o: in function `wm8731_i2c_probe': wm8731.c:(.text+0x609): undefined reference to `__devm_regmap_init_i2c' ld: sound/soc/codecs/wm8731.o: in function `wm8731_modinit': wm8731.c:(.init.text+0x8): undefined reference to `i2c_register_driver' ld: sound/soc/codecs/wm8731.o: in function `wm8731_exit': wm8731.c:(.exit.text+0x6): undefined reference to `i2c_del_driver' but CONFIG_I2C=m. Full randconfig file is attached. Note: I only see the kconfig warnings when I build for i386 on an x86_64 machine. If I build on an i386 system, I don't see those warnings, but I do see the linker errors. -- ~Randy

1 0

[alsa-devel] [PATCH v2] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 25 Sep '18

25 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- v2: Removed duplicate MIC lines. --- sound/soc/codecs/cs4265.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..4e55ac46 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"LINEOUTR", NULL, "DAC"}, {"SPDIFOUT", NULL, "SPDIF"}, + {"Pre-amp MIC", NULL, "MICL"}, + {"Pre-amp MIC", NULL, "MICR"}, + {"ADC Mux", "MIC", "Pre-amp MIC"}, {"ADC Mux", "LINEIN", "LINEINL"}, {"ADC Mux", "LINEIN", "LINEINR"}, - {"ADC Mux", "MIC", "MICL"}, - {"ADC Mux", "MIC", "MICR"}, {"ADC", NULL, "ADC Mux"}, {"DOUT", NULL, "ADC"}, {"DAI1 Capture", NULL, "DOUT"}, -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_ops
by Matthias Kaehlcke 25 Sep '18

25 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..ab44133bc0bf 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

3 2

[alsa-devel] [PATCH - Fix typo in configure.ac 1/1] PATCH - Fix typo in configure.ac
by plevine457＠gmail.com 25 Sep '18

25 Sep '18

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/configure.ac b/configure.ac index 3e2d233..cc3740b 100644 --- a/configure.ac +++ b/configure.ac @@ -90,7 +90,7 @@ if test "$use_maemo_rm" = "yes"; then fi AC_ARG_ENABLE([libav], - AS_HELP_STRING([--disable-avlib], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) + AS_HELP_STRING([--disable-libav], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) if test "x$enable_libav" != "xno"; then PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no]) -- 2.19.0

2 1

Re: [alsa-devel] [PATCH 1/3] ARM: configs: omap2plus: Add tlv320aic23 as module
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > The AM3517 EVM has an expander board which has two tlv320aic23 > codecs. This enables the driver for these codecs as a module. Applying into omap-for-v4.20/defconfig thanks. Tony

1 0

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. Mark do you want to pick this one? If so: Acked-by: Tony Lindgren <tony(a)atomide.com> Maybe update the subject line according too. Regards, Tony

1 0

Re: [alsa-devel] [PATCH 2/3] ARM: DTS: AM3517-EVM: Add support for UI board and Audio
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 15:41]: > The full AM3517-EVM shipped with an add-on board which contained > two Audio codecs, a GPIO expander with a variety of buttons, and > some other features. This patch enables these portions of the UI > board because they don't directly conflict with existing features. Applying this one into omap-for-v4.20/dt thanks. Tony

1 0

[alsa-devel] [PATCH] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 24 Sep '18

24 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..332d714a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -230,6 +230,10 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"DAI1 Capture", NULL, "DOUT"}, {"DAI2 Capture", NULL, "DOUT"}, + { "Pre-amp MIC", NULL, "MICL" }, + { "Pre-amp MIC", NULL, "MICR" }, + { "ADC Mux", "MIC", "Pre-amp MIC" }, + /* Loopback */ {"Loopback", "Switch", "ADC"}, {"DAC", NULL, "Loopback"}, -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Jarkko Nikula 23 Sep '18

23 Sep '18

On 09/23/18 18:37, Adam Ford wrote: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. > > Signed-off-by: Adam Ford <aford173(a)gmail.com> > For sound/soc/omap/ changes: Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com>

1 0

[alsa-devel] Pinpoint audio audio corruption (Layla3G)
by Mark Hills 23 Sep '18

23 Sep '18

I have a bug in ALSA somewhere; playback is corrupted. Nothing is pro-actively telling me of a problem (eg. underrun) I could use some assistance getting to the bottom of this. Listen to [1] for what it sounds like -- analogue recording from the output. Look at the waveform [2] in Baudlien and it seems both L+R channels geting zero'd for some buffer period of 2.5 or 5 milliseconds. This is Layla3G, snd-echo3g. This problem has been rare, but persisted for many years. I've finally got something reproducable to investigate. It began ~10 years ago where very occasionally recordings or playback would do this. This was a simple self-contained application in production use; written by me, using plughw:Layla3G,0,0 (no desktops, browsers, dmix etc.) Problems were frequent enough to be annoying, but infrequent enough that I just had to put it down to 'cosmic rays', with no useful bug report. These days I am on a desktop using dmix; and more recently with Google Chrome and found I can reliably 'trigger' it. To reset, I must close all applications using /dev/snd, and re-open. It seems to be: * whether or not Google Chrome is first to open the device * about 10-60 minutes after Chrome has opened the device * more prone to happening if other applications are using the dmix * not related to buffer sizes; I use larger sizes in the asoundrc it doesn't affect the likelihood, or the 'sound' when it does happen * I've seen two Layla's affected * I've had many kernels and PCs over the years There's a bug somewhere, and with nothing printed to dmesg, console etc., I'm at a loss here, imagining my problem could be anywhere between dmix, alsa-lib, snd-echo3g or even the firmware itself. Whatever the case, I've finally got some combination I can do in userland as a trigger. Below is: - .asoundrc describing today's setup (previously I used stereo with no route, still with problems) - output of the 'aplay' that produced the recording above; and - some other stats. [1] http://www.pogo.org.uk/~mark/tmp/layla-glitch.flac [2] http://www.pogo.org.uk/~mark/tmp/layla-glitch.png Many thanks -- Mark pcm.!default { type plug slave.pcm layla_multi } pcm.layla_multi { type route slave.pcm { type dmix ipc_key 1742 slave { pcm "hw:Layla3G,0,0" period_size 992 buffer_size 10912 channels 5 rate 48000 format S16_LE } } ttable { # Headphones 0.0 1 1.1 1 # Speakers 0.2 1 1.3 1 # Subwoofer; mono mix 0.4 0.5 1.4 0.5 } } $ uname -a Linux stax 4.18.5 #113 SMP PREEMPT Mon Sep 3 23:14:59 BST 2018 x86_64 Intel(R) Core(TM) i5-6500 CPU @ 3.20GHz GenuineIntel GNU/Linux $ head -99l /proc/asound/Layla3G/pcm0p/sub0/* ==> /proc/asound/Layla3G/pcm0p/sub0/hw_params <== access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 6 rate: 48000 (48000/1) period_size: 992 buffer_size: 10912 ==> /proc/asound/Layla3G/pcm0p/sub0/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: Analog PCM name: Layla3G subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 8 subdevices_avail: 7 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc <== 128 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc_max <== 256 ==> /proc/asound/Layla3G/pcm0p/sub0/status <== state: RUNNING owner_pid : 9250 trigger_time: 663739.627559736 tstamp : 680208.555428028 delay : -790500256 avail : 790511168 avail_max : 790511168 ----- hw_ptr : 790500256 appl_ptr : 0 ==> /proc/asound/Layla3G/pcm0p/sub0/sw_params <== tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 6142909891733356544 silence_threshold: 0 silence_size: 6142909891733356544 boundary: 6142909891733356544 $ aplay -vv all48000.wav Playing WAVE 'all48000.wav' : Float 32 bit Little Endian, Rate 48000 Hz, Stereo Plug PCM: Linear Integer <-> Linear Float conversion PCM (S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 1 2 <- none 3 <- none 4 <- none 5 <- none Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM Transformation table: 0 <- 0 1 <- 1 2 <- 0 3 <- 1 4 <- 0*0.5 + 1*0.5 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Hardware PCM card 0 'Layla3G' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 appl_ptr : 0 hw_ptr : 807111136 ############################ + | 56%^C Aborted by signal Interrupt... aplay: pcm_write:2004: write error: Interrupted system call

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: ensure that cluster header size is enough
by Dan Carpenter 22 Sep '18

22 Sep '18

I think we need a check to make sure that "hc_header.wLength" is large enough for a struct struct uac3_cluster_header_descriptor. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..f7179ce39a19 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -968,6 +968,12 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * and request Cluster Descriptor */ wLength = le16_to_cpu(hc_header.wLength); + if (wLength < sizeof(*cluster)) { + dev_err(&dev->dev, + "%u:%d : cluster header size too small %d\n", + iface_no, altno, wLength); + return ERR_PTR(-EIO); + } cluster = kzalloc(wLength, GFP_KERNEL); if (!cluster) return ERR_PTR(-ENOMEM);

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

Hi Takashi, Please hold applying the patch, I'm having final checks with RME regarding the VID or VID:PID based approach when to use this functionality. If it seems better to use strict VID:PID table instead, I will send an updated patch. - Jussi

1 0

[alsa-devel] [RFC] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

This patch accessing vendor specific registers of RME Class Compliant devices works otherwise fine, but there is still one small problem. It is supposed to be a polled from the hardware, since the values indicate device status and are subject to change at all times. Behavior is correct if I keep running "amixer" all the time to retrieve values - sampling rate changes are visible and values get updated. However, when I poll the values from an application without reopening the mixer, they are being cached somewhere and the changes are not reflected. This problem doesn't exist for example with PCIe based RME HDSPe AIO and application gets updated values, but in this USB based device case they don't get updated. I have flagged all the mixer controls as read-only and volatile, but the volatile flag is not honored in this (assuming I've understood it correctly). Any guidance on how to fix the caching problem is welcome!

3 10

[alsa-devel] [PATCH 0/2] ASoC: for_each_component_xx() macro
by Kuninori Morimoto 21 Sep '18

21 Sep '18

Hi Mark These are v2 of for_each_component_xx() macro. v1 -> v2 is renamed comment_all to comment. Kuninori Morimoto (2): ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 35 ++++++++++++++++++++--------------- 2 files changed, 25 insertions(+), 15 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI clock during suspend/resume modes
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Prepare <-> Cleanup functions pair has balanced calls. But in case of suspend mode no call to rsnd_soc_dai_shutdown() function, so cleanup isn't called. OTOH during resume mode function rsnd_soc_dai_prepare() is called, but calling rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero, bacause was not cleanup before). We need to call rsnd_ssi_prepare(), because it enables SSI clocks by calling rsnd_ssi_master_clk_start(). This patch allows to call prepare/cleanup functions always. Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> [kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 7 +++---- sound/soc/sh/rcar/rsnd.h | 14 +++++++------- 2 files changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0bbc4b0..6d19475 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -134,10 +134,9 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e857311..4464d1d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -318,9 +318,8 @@ struct rsnd_mod { /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -331,9 +330,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_prepare 0 -#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -345,11 +343,13 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ +#define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_prepare 1 -#define __rsnd_mod_add_cleanup -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,7 +363,7 @@ struct rsnd_mod { #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 #define __rsnd_mod_call_prepare 0 -#define __rsnd_mod_call_cleanup 1 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: tidyup for_each_card_prelinks() dai_link
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 7fe072b4df5d0 ("ASoC: add for_each_card_prelinks() macro") added new for_each_card_prelinks() macro, but it had typo. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 ++++---- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 875f846..97f9f38 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -311,9 +311,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -326,9 +326,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index c67f629..6bc1d3d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -107,9 +107,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -122,9 +122,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index ff2e0ca..cc41eb5 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -172,7 +172,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) for_each_card_prelinks(card, i, dai_link) { if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -183,9 +183,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include from q6adm.c" to the asoc tree
by Mark Brown 21 Sep '18

21 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include from q6adm.c 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 c78d42c7fbd6f2a27a665bcd1ab60c9df617f7c9 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Fri, 21 Sep 2018 18:24:58 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include from q6adm.c We include wait.h twice in q6adm.c. it is unnecessary. hence remove it. Further, order the include files as alphabet. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6adm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include <linux/slab.h> -#include <linux/wait.h> -#include <linux/kernel.h> #include <linux/device.h> -#include <linux/module.h> -#include <linux/sched.h> #include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/kref.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/kref.h> -#include <linux/wait.h> -#include <linux/soc/qcom/apr.h> #include <linux/platform_device.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/soc/qcom/apr.h> +#include <linux/wait.h> #include <sound/asound.h> #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 -- 2.19.0

1 0

[alsa-devel] [PATCH] ASoC: fix a use after free case
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

Unloading ASoC modules as used by the SOF driver leads to an object being used after it's been freed. Fix this be clearing a reference to it and making sure to check for its presence. This fixes issue #144. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-core.c | 6 +++--- sound/soc/soc-topology.c | 5 +++++ 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..dce9d53ff283 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,9 +942,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { + if (dai && dai->probed) { + if (dai->driver && dai->driver->remove_order == order && + dai->driver->remove) { err = dai->driver->remove(dai); if (err < 0) dev_err(dai->dev, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..6d438ec8c0ec 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); -- 2.17.1

1 0

[alsa-devel] [PATCH] ASoC:topology:bug fix:check return value avoid oops.
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

From: Wu Zhigang <zhigang.wu(a)linux.intel.com> check the return value to free the kcontrols instance to avoid oops caused by the pointer dereference. Signed-off-by: Wu Zhigang <zhigang.wu(a)linux.intel.com> [guennadi.liakhovetski(a)linux.intel.com add handling of .autodisable=1 cases] Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-dapm.c | 99 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 90 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..8dc31a7a42fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2402,12 +2402,11 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } -void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +static void snd_soc_dapm_free_widget_data(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *p, *next_p; enum snd_soc_dapm_direction dir; - list_del(&w->list); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2420,6 +2419,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w->kcontrols); kfree_const(w->name); +} + +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + list_del(&w->list); + snd_soc_dapm_free_widget_data(w); kfree(w); } @@ -3038,11 +3043,16 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w, *last; unsigned int val; + int ret = 0; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + /* + * widgets with the snd_soc_dapm_kcontrol ID and .num_controls = 0 can + * be appended to the list while scanning it, this is safe. + */ list_for_each_entry(w, &card->widgets, list) { if (w->new) @@ -3053,8 +3063,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&card->dapm_mutex); - return -ENOMEM; + ret = -ENOMEM; + goto out_free; } } @@ -3062,23 +3072,29 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - dapm_new_mixer(w); + ret = dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - dapm_new_mux(w); + ret = dapm_new_mux(w); break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - dapm_new_pga(w); + ret = dapm_new_pga(w); break; case snd_soc_dapm_dai_link: - dapm_new_dai_link(w); + ret = dapm_new_dai_link(w); break; default: break; } + if (ret < 0) { + kfree(w->kcontrols); + w->kcontrols = NULL; + goto out_free; + } + /* Read the initial power state from the device */ if (w->reg >= 0) { soc_dapm_read(w->dapm, w->reg, &val); @@ -3087,6 +3103,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) if (val == w->on_val) w->power = 1; } + } + + list_for_each_entry(w, &card->widgets, list) { + if (w->new) + continue; w->new = 1; @@ -3097,6 +3118,66 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&card->dapm_mutex); return 0; + +out_free: + last = w; + + /* + * If any new widgets have been created above for .autodisable = 1 + * controls, they are also on this list, but at its very end. We're + * processing an error case, so the loop above was interrupted before + * reaching dynamically added widgets, since the latter cannot fail - + * their .num_controls = 0, so allocation cannot fail, and their type is + * snd_soc_dapm_kcontrol, those cannot fail either. Therefore "last" + * points to a widget before the first dynamic one. + */ + list_for_each_entry(w, &card->widgets, list) { + unsigned int i; + + if (w->new) + continue; + + if (w == last) + break; + + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_mixer_control *mc = + (struct soc_mixer_control *) + kcontrol->private_value; + + if (mc->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_enum *e = (struct soc_enum *) + kcontrol->private_value; + + if (e->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + default: + break; + } + + snd_soc_dapm_free_widget_data(w); + } + + mutex_unlock(&card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -- 2.17.1

1 0

[alsa-devel] [PATCH v2] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 21 Sep '18

21 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to increase readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- Changes v2: - Fix typo in commit description, ncrease to increase. sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

2 3

[alsa-devel] [PATCH 00/11] add add for_each_xx() macro
by Kuninori Morimoto 20 Sep '18

20 Sep '18

Hi Mark Current ALSA SoC is using many list_for_each_entry() and/or for() loop for each lists. But, these need to use same pattern, and it is not readable. Now, "codec list loop for rtd", and "codec list loop for dai_link" had been replaced to for_each_xxx() macro. These patches replace "xxx list loop for yyy" by using for_each_yyy_xxx() macro. # Maybe some of them are overkill replacement, and/or # using not good naming. I'm happy to get feedback 1) - 2) : fixup previous for_each_xxx() macro patch 3) - 11) : new for_each_xxx() macro Kuninori Morimoto (11): 1) ASoC: convert for_each_rtd_codec_dai() for missing part 2) ASoC: rename for_each_rtd_codec_dai_reverse to rollback 3) ASoC: add for_each_card_prelinks() macro 4) ASoC: add for_each_card_links() macro 5) ASoC: add for_each_card_rtds() macro 6) ASoC: add for_each_card_components() macro 7) ASoC: add for_each_component_all() macro 8) ASoC: add for_each_component_dais() macro 9) ASoC: add for_each_comp_order() macro 10) ASoC: add for_each_dpcm_fe() macro 11) ASoC: add for_each_dpcm_be() macro include/sound/soc-dpcm.h | 10 ++ include/sound/soc.h | 29 ++++- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 5 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 13 +-- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 7 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/samsung/tm2_wm5110.c | 13 +-- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 130 +++++++++++------------ sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 101 +++++++++--------- 29 files changed, 230 insertions(+), 193 deletions(-) -- 2.7.4

2 21

[alsa-devel] Applied "ASoC: rt5663: Remove the boost volume in the beginning of playback" to the asoc tree
by Mark Brown 20 Sep '18

20 Sep '18

The patch ASoC: rt5663: Remove the boost volume in the beginning of playback 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 fc795bf7224efda8c35e505966c2757856064247 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 19 Sep 2018 09:56:47 +0800 Subject: [PATCH] ASoC: rt5663: Remove the boost volume in the beginning of playback The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..2444fad7c2df 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.19.0

1 0

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

20 Sep '18

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

2 22

[alsa-devel] SoftVol in user space
by Riccardo Casagrande 20 Sep '18

20 Sep '18

Hello devs, I have to add a volume control to my custom card (It has an i.MX6 chip and 3 FDA2100 from TI as audio chip) because the FDAs have no volume controller. This below is the .asoundrc file I made. It works great if I manually made a speaker-test of one of devicefdaX sound card. Then I can see on alsamixer the volume control, but my need is to access and control softvol from user space. pcm.softvolfda0 { type softvol slave { pcm "hw:0,0" } control { name "volume0" card 0 device 0 } } pcm.softvolfda1 { type softvol slave { pcm "hw:0,1" } control { name "volume1" card 0 device 1 } } pcm.softvolfda2 { type softvol slave { pcm "hw:0,2" } control { name "volume2" card 0 device 2 } } pcm.devicefda0 { type plug slave.pcm "softvolfda0" } pcm.devicefda1 { type plug slave.pcm "softvolfda1" } pcm.devicefda2 { type plug slave.pcm "softvolfda2" } I saw that on alsalib there is this command int snd_pcm_softvol_open (snd_pcm_t ** pcmp, const char * name, snd_pcm_format_t sformat, int ctl_card, snd_ctl_elem_id_t * ctl_id, int cchannels, double min_dB, double max_dB, int resolution, snd_pcm_t * slave, int close_slave ) but I can't find any example of usage.This is the log of the error: mp && slave' failed. Aborted Any suggestion is welcome Riccardo Casagrande

1 0

[alsa-devel] [PATCH 00/15] Add support for Sound BlasterX AE-5
by Connor McAdams 20 Sep '18

20 Sep '18

This patch series adds support for the Sound BlasterX AE-5. I have personally tested it, and had others test it as well. All outputs work, front headphone, rear headphone, rear line-out, surround, digital. The only issue currently is that the microphone suffers from inconsistency like the Sound Blaster Z does, and Creative may be willing to help fix that, but if they're not, I have a few ideas. I also have the ZxR working currently. That will require some more complicated changes of the driver, so I have left it out and just focused on the AE-5. Connor McAdams (15): ALSA: hda/ca0132 - Define new verbs and control params ALSA: hda/ca0132 - Add quirk for Sound BlasterX AE-5 ALSA: hda/ca0132 - Add pincfg for AE-5 ALSA: hda/ca0132 - Set AE-5 bools and select mixer ALSA: hda/ca0132 - Change ca0132_mmio_init for AE-5 ALSA: hda/ca0132 - Add AE-5 pre-init and ca0113 functions ALSA: hda/ca0132 - Add AE-5 regular init setup ALSA: hda/ca0132 - Change firmware name and usage ALSA: hda/ca0132 - Merge post-dsp functions + cleanup ALSA: hda/ca0132 - Add DSP setup functions for AE-5 ALSA: hda/ca0132 - Clean up ca0132_alt_out_select ALSA: hda/ca0132 - Add output set commands for AE-5 ALSA: hda/ca0132 - Add input selection commands for AE-5 ALSA: hda/ca0132 - Add AE-5 specific controls ALSA: hda/ca0132 - Add AE-5 exit function sound/pci/hda/patch_ca0132.c | 1283 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 1038 insertions(+), 245 deletions(-) -- 2.7.4

2 17

[alsa-devel] [PATCH - hwmixvolume v2 1/7] hwmixvolume: use four spaces instead of one tab for indent
by Emmanuel Gil Peyrot 20 Sep '18

20 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This is the recommended coding style for all Python programs, as specified in PEP-0008[1]. [1] https://www.python.org/dev/peps/pep-0008/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index ef80bc8..038bcb3 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -26,285 +26,285 @@ EVENT_INFO = alsahcontrol.event_mask['INFO'] EVENT_REMOVE = alsahcontrol.event_mask_remove class Stream: - def __init__(self, element, parent): - self.element = element - self.element.set_callback(self) - self.parent = parent - self.label = None - self.scales = [] - self.adjustments = [] - self.callback(self.element, EVENT_INFO) - - def destroy(self): - self.deactivate() - - def callback(self, e, mask): - if mask == EVENT_REMOVE: - self.deactivate() - elif (mask & EVENT_INFO) != 0: - info = alsahcontrol.Info(self.element) - if info.is_inactive: - self.deactivate() - else: - self.activate() - elif (mask & EVENT_VALUE) != 0: - self.update_scales_from_ctl() - - def activate(self): - if self.label: - return - info = alsahcontrol.Info(self.element) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) - self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) - for i in range(info.count): - adj = gtk.Adjustment(value=values[i], - lower=info.min, upper=info.max, - step_incr=1, - page_incr=(info.max-info.min+1)/8) - adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) - scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) - self.scales.append(scale) - self.adjustments.append(adj) - self.parent.scales_vbox.show_all() - self.parent.update_msg_label() - - def deactivate(self): - if not self.label: - return - self.label.destroy() - for s in self.scales: - s.destroy() - self.label = None - self.scales = [] - self.adjustments = [] - self.parent.update_msg_label() - - def update_scales_from_ctl(self): - if not self.label: - return - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, count) - for i in range(count): - self.adjustments[i].set_value(values[i]) - - def update_ctl_from_scale(self, adj, index): - scale_value = adj.get_value() - value_to_set = int(round(adj.get_value())) - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - if self.parent.lock_check.get_active(): - values = [value_to_set for i in range(count)] - else: - value.read() - values = value.get_array(TYPE_INTEGER, count) - values[index] = value_to_set - value.set_array(TYPE_INTEGER, values) - value.write() - if value_to_set != scale_value: - adj.set_value(value_to_set) - - def get_label(self, info): - pid = self.get_pid(info) - if pid: - cmdline = self.get_pid_cmdline(pid) - if cmdline: - return cmdline - else: - return "PID %d" % pid - else: - name = info.name - if name[-7:] == " Volume": - name = name[:-7] - if name[-9:] == " Playback": - name = name[:-9] - return name - - def get_pid(self, info): - card = self.parent.current_card - device = info.device - subdevice = info.subdevice - if subdevice == 0: - subdevice = info.index - filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) - try: - f = open(filename, "r") - except IOError: - return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() - return None - - def get_pid_cmdline(self, pid): - try: - f = open("/proc/%d/cmdline" % pid, "r") - except IOError: - return None - try: - cmdline = f.read() - finally: - f.close() - return cmdline.replace('\x00', ' ').strip() + def __init__(self, element, parent): + self.element = element + self.element.set_callback(self) + self.parent = parent + self.label = None + self.scales = [] + self.adjustments = [] + self.callback(self.element, EVENT_INFO) + + def destroy(self): + self.deactivate() + + def callback(self, e, mask): + if mask == EVENT_REMOVE: + self.deactivate() + elif (mask & EVENT_INFO) != 0: + info = alsahcontrol.Info(self.element) + if info.is_inactive: + self.deactivate() + else: + self.activate() + elif (mask & EVENT_VALUE) != 0: + self.update_scales_from_ctl() + + def activate(self): + if self.label: + return + info = alsahcontrol.Info(self.element) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, info.count) + self.label = gtk.Label(self.get_label(info)) + self.label.set_single_line_mode(True) + self.parent.scales_vbox.pack_start(self.label, expand=False) + for i in range(info.count): + adj = gtk.Adjustment(value=values[i], + lower=info.min, upper=info.max, + step_incr=1, + page_incr=(info.max-info.min+1)/8) + adj.connect('value-changed', self.update_ctl_from_scale, i) + scale = gtk.HScale(adj) + scale.set_draw_value(False) + self.parent.scales_vbox.pack_start(scale, expand=False) + self.scales.append(scale) + self.adjustments.append(adj) + self.parent.scales_vbox.show_all() + self.parent.update_msg_label() + + def deactivate(self): + if not self.label: + return + self.label.destroy() + for s in self.scales: + s.destroy() + self.label = None + self.scales = [] + self.adjustments = [] + self.parent.update_msg_label() + + def update_scales_from_ctl(self): + if not self.label: + return + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, count) + for i in range(count): + self.adjustments[i].set_value(values[i]) + + def update_ctl_from_scale(self, adj, index): + scale_value = adj.get_value() + value_to_set = int(round(adj.get_value())) + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + if self.parent.lock_check.get_active(): + values = [value_to_set for i in range(count)] + else: + value.read() + values = value.get_array(TYPE_INTEGER, count) + values[index] = value_to_set + value.set_array(TYPE_INTEGER, values) + value.write() + if value_to_set != scale_value: + adj.set_value(value_to_set) + + def get_label(self, info): + pid = self.get_pid(info) + if pid: + cmdline = self.get_pid_cmdline(pid) + if cmdline: + return cmdline + else: + return "PID %d" % pid + else: + name = info.name + if name[-7:] == " Volume": + name = name[:-7] + if name[-9:] == " Playback": + name = name[:-9] + return name + + def get_pid(self, info): + card = self.parent.current_card + device = info.device + subdevice = info.subdevice + if subdevice == 0: + subdevice = info.index + filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) + try: + f = open(filename, "r") + except IOError: + return None + try: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) + finally: + f.close() + return None + + def get_pid_cmdline(self, pid): + try: + f = open("/proc/%d/cmdline" % pid, "r") + except IOError: + return None + try: + cmdline = f.read() + finally: + f.close() + return cmdline.replace('\x00', ' ').strip() class MixerWindow(gtk.Window): - card_numbers = alsacard.card_list() - current_card = -1 - hcontrol = None - scales_vbox = None - msg_label = None - streams = [] - hctl_sources = [] - - def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) - self.set_title("Hardware Mixer Volumes") - - vbox = gtk.VBox() - self.add(vbox) - - hbox = gtk.HBox() - vbox.pack_start(hbox, expand=False) - - label = gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) - - combo = gtk.combo_box_new_text() - for i in self.card_numbers: - str = "%d: %s" % (i, alsacard.card_get_name(i)) - combo.append_text(str) - if len(self.card_numbers) > 0: - combo.set_active(0) - combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) - label.set_mnemonic_widget(combo) - - self.lock_check = gtk.CheckButton(label="_Lock Channels") - self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) - - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) - - self.scales_vbox = gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) - - label = gtk.Label() - label.set_single_line_mode(True) - line_height = label.size_request()[1] - label.destroy() - scale = gtk.HScale() - scale.set_draw_value(False) - line_height += scale.size_request()[1] - scale.destroy() - # always have space for at least four sliders - scrollwin.set_size_request(width=-1, height=line_height*4+4) - - # TODO: select the default card or the first card with stream controls - if len(self.card_numbers) > 0: - self.change_card(self.card_numbers[0]) - self.update_msg_label() - - self.show_all() - - def change_card(self, cardnum): - for s in self.hctl_sources: - gobject.source_remove(s) - self.hctl_sources = [] - - self.hcontrol = self.open_hcontrol_for_card(cardnum) - - for s in self.streams: - s.destroy() - self.streams = [] - - self.current_card = cardnum - - if not self.hcontrol: - self.update_msg_label() - return - - for id in self.hcontrol.list(): - if not self.is_stream_elem(id): - continue - elem = alsahcontrol.Element(self.hcontrol, id[0]) - info = alsahcontrol.Info(elem) - if not self.is_stream_info(info): - continue - stream = Stream(elem, self) - self.streams.append(stream) - - for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) - - self.update_msg_label() - - self.scales_vbox.show_all() - - def update_msg_label(self): - needs_msg = len(self.scales_vbox.get_children()) < 2 - has_msg = self.msg_label - if has_msg and not needs_msg: - self.msg_label.destroy() - self.msg_label = None - elif needs_msg: - if len(self.streams) > 0: - msg = "There are no open streams." - else: - msg = "This card does not have stream controls." - if not has_msg: - self.msg_label = gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) - self.scales_vbox.show_all() - elif self.msg_label.get_text() != msg: - self.msg_label.set_text(msg) - - def open_hcontrol_for_card(self, cardnum): - devname = "hw:CARD=" + str(cardnum) - try: - hc = alsahcontrol.HControl(name=devname, - mode=alsahcontrol.open_mode['NONBLOCK']) - except: - # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, - "Cannot open sound card control device.") - dlg.run() - dlg.destroy() - return None - return hc - - def is_stream_elem(self, id): - return ((id[1] == INTF_PCM and - id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or - (id[1] == INTF_MIXER and - id[4] == "VIA DXS Playback Volume")) - - def is_stream_info(self, info): - return info.is_readable and info.is_writable and info.type == TYPE_INTEGER - - def hctl_io_callback(self, source, condition): - self.hcontrol.handle_events() - return True + card_numbers = alsacard.card_list() + current_card = -1 + hcontrol = None + scales_vbox = None + msg_label = None + streams = [] + hctl_sources = [] + + def __init__(self): + gtk.Window.__init__(self) + self.connect('destroy', lambda w: gtk.main_quit()) + self.set_title("Hardware Mixer Volumes") + + vbox = gtk.VBox() + self.add(vbox) + + hbox = gtk.HBox() + vbox.pack_start(hbox, expand=False) + + label = gtk.Label("_Sound Card: ") + label.set_use_underline(True) + hbox.pack_start(label, expand=False) + + combo = gtk.combo_box_new_text() + for i in self.card_numbers: + str = "%d: %s" % (i, alsacard.card_get_name(i)) + combo.append_text(str) + if len(self.card_numbers) > 0: + combo.set_active(0) + combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) + hbox.pack_start(combo) + label.set_mnemonic_widget(combo) + + self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check.set_active(True) + vbox.pack_start(self.lock_check, expand=False) + + scrollwin = gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(gtk.SHADOW_NONE) + vbox.pack_start(scrollwin) + + self.scales_vbox = gtk.VBox() + scrollwin.add_with_viewport(self.scales_vbox) + + label = gtk.Label() + label.set_single_line_mode(True) + line_height = label.size_request()[1] + label.destroy() + scale = gtk.HScale() + scale.set_draw_value(False) + line_height += scale.size_request()[1] + scale.destroy() + # always have space for at least four sliders + scrollwin.set_size_request(width=-1, height=line_height*4+4) + + # TODO: select the default card or the first card with stream controls + if len(self.card_numbers) > 0: + self.change_card(self.card_numbers[0]) + self.update_msg_label() + + self.show_all() + + def change_card(self, cardnum): + for s in self.hctl_sources: + gobject.source_remove(s) + self.hctl_sources = [] + + self.hcontrol = self.open_hcontrol_for_card(cardnum) + + for s in self.streams: + s.destroy() + self.streams = [] + + self.current_card = cardnum + + if not self.hcontrol: + self.update_msg_label() + return + + for id in self.hcontrol.list(): + if not self.is_stream_elem(id): + continue + elem = alsahcontrol.Element(self.hcontrol, id[0]) + info = alsahcontrol.Info(elem) + if not self.is_stream_info(info): + continue + stream = Stream(elem, self) + self.streams.append(stream) + + for fd,condition in self.hcontrol.poll_fds: + self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + + self.update_msg_label() + + self.scales_vbox.show_all() + + def update_msg_label(self): + needs_msg = len(self.scales_vbox.get_children()) < 2 + has_msg = self.msg_label + if has_msg and not needs_msg: + self.msg_label.destroy() + self.msg_label = None + elif needs_msg: + if len(self.streams) > 0: + msg = "There are no open streams." + else: + msg = "This card does not have stream controls." + if not has_msg: + self.msg_label = gtk.Label(msg) + self.scales_vbox.pack_start(self.msg_label) + self.scales_vbox.show_all() + elif self.msg_label.get_text() != msg: + self.msg_label.set_text(msg) + + def open_hcontrol_for_card(self, cardnum): + devname = "hw:CARD=" + str(cardnum) + try: + hc = alsahcontrol.HControl(name=devname, + mode=alsahcontrol.open_mode['NONBLOCK']) + except: + # TODO: alsa error msg + dlg = gtk.MessageDialog(self, + gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, + gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + "Cannot open sound card control device.") + dlg.run() + dlg.destroy() + return None + return hc + + def is_stream_elem(self, id): + return ((id[1] == INTF_PCM and + id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or + (id[1] == INTF_MIXER and + id[4] == "VIA DXS Playback Volume")) + + def is_stream_info(self, info): + return info.is_readable and info.is_writable and info.type == TYPE_INTEGER + + def hctl_io_callback(self, source, condition): + self.hcontrol.handle_events() + return True def main(): - MixerWindow() - gtk.main() + MixerWindow() + gtk.main() main() -- 2.19.0

3 5

[alsa-devel] [PATCH] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 20 Sep '18

20 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to ncrease readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: q6afe: dt-bindings: Update input range for qcom, sd-lines
by Rohit kumar 19 Sep '18

19 Sep '18

Input to qcom,sd-lines should be between 0 and 3 instead of 1 to 4 as 0 corresponds to BIT(0) which is MI2S_SD0 line. Bit 1 to 3 corresponds to SD1 to SD3 lines respectively. Updated documentation for the same. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index a817940..d74888b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties. Usage: required for mi2s interface Value type: <prop-encoded-array> Definition: Must be list of serial data lines used by this dai. - should be one or more of the 1-4 sd lines. + should be one or more of the 0-3 sd lines. - qcom,tdm-sync-mode: Usage: required for tdm interface @@ -137,42 +137,42 @@ q6afe@4 { prim-mi2s-rx@16 { reg = <16>; - qcom,sd-lines = <1 3>; + qcom,sd-lines = <0 2>; }; prim-mi2s-tx@17 { reg = <17>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; sec-mi2s-rx@18 { reg = <18>; - qcom,sd-lines = <1 4>; + qcom,sd-lines = <0 3>; }; sec-mi2s-tx@19 { reg = <19>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; tert-mi2s-rx@20 { reg = <20>; - qcom,sd-lines = <2 4>; + qcom,sd-lines = <1 3>; }; tert-mi2s-tx@21 { reg = <21>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-rx@22 { reg = <22>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-tx@23 { reg = <23>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; }; }; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

Re: [alsa-devel] [PATCH - hwmixvolume v2 0/7] Migrate to GTK+ 3.0 and allow python3
by Takashi Sakamoto 19 Sep '18

19 Sep '18

Hi, On Sep 18 2018 22:42, Emmanuel Gil Peyrot wrote: > From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> > > hwmixvolume was one of the last few programs on my system to depend on > PyGTK. This series ports it to GTK+ 3.x using gobject-introspection, > which allows it to run on Python 3.x too. > > My audio card does not have stream controls so I’ve been unable to test > whether the functionality is working properly, but from my reading of > the documentation it should be fine. > > Changes since v1: > - Reordered tab→space patch before everything else, to make it possible > to merge it as is without any other change. > - Merged fixup commits into the main switch patch, as they had no reason > to stand by themselves. > - Replaced iteration on file.readlines() with file.__iter__() directly > (thanks Sakamoto-san). > - Fix a typo in the README (thanks Sakamoto-san). > - Explicit more the reason for each patch (thanks Iwai-san), except for > trivial ones. > > Emmanuel Gil Peyrot (7): > hwmixvolume: use four spaces instead of one tab for indent > hwmixvolume: replace PyGTK with gobject-introspection > hwmixvolume: switch to GTK+ 3.0 and GLib 2.0 > hwmixvolume: use a with context to open files > hwmixvolume: switch the shebang to python > hwmixvolume: add my copyright > hwmixvolume: mention the new dependencies in the README > > hwmixvolume/README | 3 +- > hwmixvolume/hwmixvolume | 560 ++++++++++++++++++++-------------------- > 2 files changed, 283 insertions(+), 280 deletions(-) I reviewed all of the above patches. Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Thanks Takashi Sakamoto

1 0

[alsa-devel] [PATCH - hwmixvolume v2 3/7] hwmixvolume: switch to GTK+ 3.0 and GLib 2.0
by Emmanuel Gil Peyrot 19 Sep '18

19 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This replaces VBox and HBox with Grid (using Gtk.Orientation), HScale with Scale, creates labels with mnemonics, set hexpand and vexpand properly, use the correct enum container classes, use the correct getter for size request, and finally update to the correct GLib watch function. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 64d232c..8e0b6b8 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -60,18 +60,18 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = Gtk.Label(self.get_label(info)) + self.label = Gtk.Label.new(self.get_label(info)) self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) + self.parent.scales_vbox.add(self.label) for i in range(info.count): adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = Gtk.HScale(adj) + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL, adjustment=adj) scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) + self.parent.scales_vbox.add(scale) self.scales.append(scale) self.adjustments.append(adj) self.parent.scales_vbox.show_all() @@ -174,45 +174,48 @@ class MixerWindow(Gtk.Window): self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = Gtk.VBox() + vbox = Gtk.Grid() + vbox.set_orientation(Gtk.Orientation.VERTICAL) self.add(vbox) - hbox = Gtk.HBox() - vbox.pack_start(hbox, expand=False) + hbox = Gtk.Grid() + vbox.add(hbox) - label = Gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) + label = Gtk.Label.new_with_mnemonic("_Sound Card: ") + hbox.add(label) - combo = Gtk.combo_box_new_text() + combo = Gtk.ComboBoxText() + combo.set_hexpand(True) for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) if len(self.card_numbers) > 0: combo.set_active(0) combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) + hbox.add(combo) label.set_mnemonic_widget(combo) - self.lock_check = Gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) + vbox.add(self.lock_check) scrollwin = Gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(Gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) + scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) + scrollwin.set_shadow_type(Gtk.ShadowType.NONE) + scrollwin.set_vexpand(True) + vbox.add(scrollwin) - self.scales_vbox = Gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) + self.scales_vbox = Gtk.Grid() + self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) + scrollwin.add(self.scales_vbox) label = Gtk.Label() label.set_single_line_mode(True) - line_height = label.size_request()[1] + line_height = max(label.get_size_request().height, 0) label.destroy() - scale = Gtk.HScale() + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL) scale.set_draw_value(False) - line_height += scale.size_request()[1] + line_height += max(scale.get_size_request().height, 0) scale.destroy() # always have space for at least four sliders scrollwin.set_size_request(width=-1, height=line_height*4+4) @@ -252,7 +255,7 @@ class MixerWindow(Gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) self.update_msg_label() @@ -270,8 +273,9 @@ class MixerWindow(Gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = Gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) + self.msg_label = Gtk.Label.new(msg) + self.msg_label.set_vexpand(True) + self.scales_vbox.add(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: self.msg_label.set_text(msg) @@ -284,8 +288,8 @@ class MixerWindow(Gtk.Window): except: # TODO: alsa error msg dlg = Gtk.MessageDialog(self, - Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, - Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, + Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, + Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() -- 2.19.0

2 3

[alsa-devel] [PATCH 2/2] sound: soc: wm8903: Cut gpio_base data
by Linus Walleij 19 Sep '18

19 Sep '18

This variable is only used inside the file, and we should never hard-code the GPIO base, so delete this. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/wm8903.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 23e43ff40ded..545512dc4c45 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -144,7 +144,6 @@ struct wm8903_platform_data { int micdet_delay; /* Delay after microphone detection (ms) */ - int gpio_base; u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ }; @@ -1876,17 +1875,12 @@ static const struct gpio_chip wm8903_template_chip = { static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; wm8903->gpio_chip.parent = wm8903->dev; - - if (pdata->gpio_base) - wm8903->gpio_chip.base = pdata->gpio_base; - else - wm8903->gpio_chip.base = -1; + wm8903->gpio_chip.base = -1; ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) -- 2.17.1

2 4

[alsa-devel] [PATCH Resend] ASoC: rt5663: Remove the boost volume in the beginning of playback
by Oder Chiou 19 Sep '18

19 Sep '18

The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad..2444fad 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.7.4

1 0

[alsa-devel] [PATCH] sound: soc: rt5677-spi: Drop unused GPIO include
by Linus Walleij 18 Sep '18

18 Sep '18

This SPI driver does not use the legacy GPIO header so just delete it. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..84501c2020c7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -18,7 +18,6 @@ #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/slab.h> -#include <linux/gpio.h> #include <linux/sched.h> #include <linux/uaccess.h> #include <linux/regulator/consumer.h> -- 2.17.1

2 2

[alsa-devel] [PATCH 1/2] sound: soc: wm8903: Pull in platform data
by Linus Walleij 18 Sep '18

18 Sep '18

No platform in the kernel is using platform data to populate the WM8903 so pull it into the source file. This is done to handle further simplifications. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- include/sound/wm8903.h | 24 ------------------------ sound/soc/codecs/wm8903.c | 24 ++++++++++++++++++++++++ 2 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index b310c5a3a958..b2ef76d5d00d 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -11,12 +11,6 @@ #ifndef __LINUX_SND_WM8903_H #define __LINUX_SND_WM8903_H -/* - * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of - * zero in platform data means "don't touch this pin". - */ -#define WM8903_GPIO_CONFIG_ZERO 0x8000 - /* * R6 (0x06) - Mic Bias Control 0 */ @@ -245,22 +239,4 @@ #define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ #define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ -#define WM8903_NUM_GPIO 5 - -struct wm8903_platform_data { - bool irq_active_low; /* Set if IRQ active low, default high */ - - /* Default register value for R6 (Mic bias), used to configure - * microphone detection. In conjunction with gpio_cfg this - * can be used to route the microphone status signals out onto - * the GPIOs for use with snd_soc_jack_add_gpios(). - */ - u16 micdet_cfg; - - int micdet_delay; /* Delay after microphone detection (ms) */ - - int gpio_base; - u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ -}; - #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 6cb3c153ba19..23e43ff40ded 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -124,6 +124,30 @@ static const char *wm8903_supply_names[WM8903_NUM_SUPPLIES] = { "DCVDD", }; +/* + * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of + * zero in platform data means "don't touch this pin". + */ +#define WM8903_GPIO_CONFIG_ZERO 0x8000 + +#define WM8903_NUM_GPIO 5 + +struct wm8903_platform_data { + bool irq_active_low; /* Set if IRQ active low, default high */ + + /* Default register value for R6 (Mic bias), used to configure + * microphone detection. In conjunction with gpio_cfg this + * can be used to route the microphone status signals out onto + * the GPIOs for use with snd_soc_jack_add_gpios(). + */ + u16 micdet_cfg; + + int micdet_delay; /* Delay after microphone detection (ms) */ + + int gpio_base; + u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ +}; + struct wm8903_priv { struct wm8903_platform_data *pdata; struct device *dev; -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: remove redundant include" to the asoc tree
by Mark Brown 18 Sep '18

18 Sep '18

The patch ASoC: remove redundant include 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 29ca7d32d7f10737e8d165fcf40fe31d44b06bee Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Tue, 18 Sep 2018 16:16:24 +0800 Subject: [PATCH] ASoC: remove redundant include module.h already contained moduleparam.h, so it is safe to remove the redundant include. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/soc-dapm.c | 1 - 4 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..b613103d801b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -10,7 +10,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c..2a3e5fbd04e4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -13,7 +13,6 @@ #include <linux/clk.h> #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 43edaf8cd276..593a11960888 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -11,7 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 43983c69f6aa..ee6b9758ec15 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -18,7 +18,6 @@ // device reopen. #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/async.h> #include <linux/delay.h> -- 2.19.0

1 0

[alsa-devel] [PATCH 2/5] ASoC: rt5682: Improve HP performance
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> We change the settings while HP power-up for better performance. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 32 +++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 14 ++++++++++++++ 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 31ec7c0..bf7d910 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1437,6 +1437,28 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); +static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); + break; + default: + return 0; + } + + return 0; +} + static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1449,8 +1471,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_LOGIC_CTRL_2, 0x0012); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x6000); - snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, - RT5682_NG2_EN_MASK, RT5682_NG2_EN); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); break; @@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, NULL, 0), + RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref1"}, {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, @@ -2616,6 +2638,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 8068140..d82a830 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1214,6 +1214,20 @@ #define RT5682_JDH_NO_PLUG (0x1 << 4) #define RT5682_JDH_PLUG (0x0 << 4) +/* Bias current control 8 (0x0111) */ +#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2) +#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2) +#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2) +#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2) +#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 -- 2.7.4

2 1

[alsa-devel] [PATCH 3/5] ASoC: rt5682: Remove HP volume control
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch removed Headphone Playback Volume control. Due to codec settings, we don't want the user to change HP analog gain. The user could use DAC1 Playback Volume control to change playback volume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index bf7d910..2725eb7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -749,7 +749,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); @@ -1108,10 +1107,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work) } static const struct snd_kcontrol_new rt5682_snd_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, - RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), - /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), -- 2.7.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: rt5682: Update calibration function
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> The ADC/DAC path should open while calibration process. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 731c6a8..31ec7c0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2451,17 +2451,22 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); + regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2478,8 +2483,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

2 1

[alsa-devel] [PATCH 4/5] ASoC: rt5682: Fix the boost volume at the begining of playback
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch fixed the boost volume at the begining of playback while DAC volume set to lower level. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2725eb7..1809966 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -68,6 +68,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {0x01c1, 0x1000}, + {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; static const struct reg_default rt5682_reg[] = { @@ -1468,6 +1469,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_CTRL_2, 0x6000); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); break; case SND_SOC_DAPM_POST_PMD: @@ -1475,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x0); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x0000); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; default: -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 18 Sep '18

18 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: fix overrun issue and missed case where bytescount = rtd->bytescount sound/soc/amd/acp-pcm-dma.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b0e245c..0ac4b5b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,16 +1035,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { period_bytes = frames_to_bytes(runtime, runtime->period_size); - dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); - if (dscr == rtd->dma_dscr_idx_1) - pos = period_bytes; - else - pos = 0; bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) + if (bytescount >= rtd->bytescount) bytescount -= rtd->bytescount; - delay = do_div(bytescount, period_bytes); - runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) { + pos = 0; + } else { + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } + if (bytescount > 0) { + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

2 1

[alsa-devel] [PATCH 5/5] ASoC: rt5682: Minor code modification
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> Minor code changes are: - improve the readability in patch list - add i2c remove function - regmap_register_patch changes to regmap_multi_reg_write Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1809966..340f904 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -67,7 +67,7 @@ struct rt5682_priv { }; static const struct reg_sequence patch_list[] = { - {0x01c1, 0x1000}, + {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; @@ -2584,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_calibrate(rt5682); - ret = regmap_register_patch(rt5682->regmap, patch_list, + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, ARRAY_SIZE(patch_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); @@ -2659,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_rt5682, + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2694,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; -- 2.7.4

2 1