November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ALSA: usb-audio: fix OOB and return value check
by Jaejoong Kim 04 Dec '17

04 Dec '17

Hi, This patch series fixes the out-of-bound error caused by the return value of usb_string(). It was descovered by KASAN. The Patch 1 is the V2 about http://www.spinics.net/lists/alsa-devel/msg69487.html Chanes in V2: - put an explicit error bail out(by Takashi iwai) Patch1 was founded by connecting the following product. http://www.lg.com/uk/lg-friends/lg-AFD-1200 I found that it only check if the return value from usb_string() is always zero while modifying OOB KASAN message. So instead of making the modifications to OOB to V2, I sent a patch series. I am sorry to break the mail thread. Thanks jaejoong Jaejoong Kim (3): ALSA: usb-audio: Fix out-of-bound error ALSA: usb-audio: Fix return value check for usb_string() ALSA: usb-audio: Add check return value for usb_string() sound/usb/mixer.c | 41 ++++++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 17 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: tfa9879: Export OF device ID as module alias
by Javier Martinez Canillas 01 Dec '17

01 Dec '17

The I2C core always reports a MODALIAS of the form i2c:<foo> even if the device was registered via OF, this means that exporting the OF device ID table device aliases in the module is not needed. But in order to change how the core reports modaliases to user-space, it's better to export it. Before this patch: $ modinfo sound/soc/codecs/snd-soc-tfa9879.ko | grep alias alias: i2c:tfa9879 After this patch: $ modinfo sound/soc/codecs/snd-soc-tfa9879.ko | grep alias alias: i2c:tfa9879 alias: of:N*T*Cnxp,tfa9879C* alias: of:N*T*Cnxp,tfa9879 Signed-off-by: Javier Martinez Canillas <javierm(a)redhat.com> --- sound/soc/codecs/tfa9879.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index f8dd67ca0744..e7ca764b5729 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -316,6 +316,7 @@ static const struct of_device_id tfa9879_of_match[] = { { .compatible = "nxp,tfa9879", }, { } }; +MODULE_DEVICE_TABLE(of, tfa9879_of_match); static struct i2c_driver tfa9879_i2c_driver = { .driver = { -- 2.14.3

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[alsa-devel] [PATCH 1/5 v5] ASoC: ep93xx-ac97: Fix platform_get_irq's error checking
by Arvind Yadav 01 Dec '17

01 Dec '17

The platform_get_irq() function returns negative if an error occurs. zero or positive number on success. platform_get_irq() error checking for zero is not correct. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- changes in v2 : irq was unsigned. so changed it to signed. changes in v3 : Add failure case '<= 0' instead of '< 0'. IRQ0 is not valid. changes in v4 : Return -ENODEV insted of irq. changes in v5 : Add separate error for irq == 0 and irq < 0. sound/soc/cirrus/ep93xx-ac97.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index bbf7a92..cd5a939 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -365,7 +365,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; struct resource *res; - unsigned int irq; + int irq; int ret; info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -378,8 +378,8 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); - if (!irq) - return -ENODEV; + if (irq <= 0) + return irq < 0 ? irq : -ENODEV; ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, IRQF_TRIGGER_HIGH, pdev->name, info); -- 2.7.4

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[alsa-devel] [PATCH 00/17] Add Headphone Detection to TLV320AIC31xx Driver
by Andrew F. Davis 01 Dec '17

01 Dec '17

Hello all, This series has the end goal of adding headphone detection to the tlv320aic31xx driver. The first few patches are mostly cleanups. Then a couple bug fixes I noticed. Followed by adding interrupt handling and finally headphone detection. The last two, as their commit name recommends, should not be taken and are included in-case someone wants to evaluate headphone detection using the EVM for this device wired to a Beaglebone Black. This series (or at least patch #5) depend on this DT fix[0]. Thanks, Andrew [0]https://www.spinics.net/lists/kernel/msg2644651.html Andrew F. Davis (17): ASoC: tlv320aic31xx: General source formatting cleanup ASoC: tlv320aic31xx: Reformat header file using GENMASK and BIT macros ASoC: tlv320aic31xx: Fix GPIO1 register definition ASoC: tlv320aic31xx: Merge init function into probe ASoC: tlv320aic31xx: Switch GPIO handling to use gpiod_* API ASoC: tlv320aic31xx: Remove platform data ASoC: tlv320aic31xx: Add MICBIAS off setting ASoC: tlv320aic31xx: Check clock and divider before division ASoC: tlv320aic31xx: Add CODEC clock slave support ASoC: tlv320aic31xx: Fix inverted BCLK handling ASoC: tlv320aic31xx: Reset registers during probe ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support ASoC: tlv320aic31xx: Add headphone/headset detection ASoC: tlv320aic31xx: Add button press detection NOT FOR MERGING: Add TLV320DAC3101 to BBB for testing NOT FOR MERGING: Add demo jack detection policy for testing .../devicetree/bindings/sound/tlv320aic31xx.txt | 1 + arch/arm/boot/dts/am335x-boneblack.dts | 106 +++++ include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 400 +++++++++++----- sound/soc/codecs/tlv320aic31xx.h | 504 ++++++++++----------- 5 files changed, 623 insertions(+), 389 deletions(-) rewrite sound/soc/codecs/tlv320aic31xx.h (83%) -- 2.15.0

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Re: [alsa-devel] [PATCH RESEND 1/2] ASoC: Intel: kbl: Remove option of choosing CH count based on pdata
by Mark Brown 01 Dec '17

01 Dec '17

On Thu, Nov 30, 2017 at 11:44:43AM +0530, Naveen M wrote: > Regarding the Query: > As Vinod mentioned, I'm working on Chrome OS user-space stack that uses > customized audio server "CRAS". > These patches are verified on a board that has only 2 dmics. > > Giving the control to CRAS, enables to select the desired dmic using > channel-map configuration in user-space. > Ex:Rear/Front mic selection for World & Front facing camera scenario People do run non-ChromeOS installs on their Chromebooks so we can't just assume that there's nobody out there with a regular distro on this hardware - a user like this reported the problems with the ABI for topology files breaking for example.

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[alsa-devel] [PATCH v4 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 01 Dec '17

01 Dec '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Adapt format name to changes in the first patch from this series. Changes from v2, v3: None. sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..c350117c8e31 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

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[alsa-devel] [PATCH v2] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 01 Dec '17

01 Dec '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - add more explain on log area - call dummy write after 5LRCK sound/soc/codecs/ak4613.c | 59 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 59 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..937cff4 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct delayed_work dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,64 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work.work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned long delay; + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delayed_work call from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + */ + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + break; + default: + return 0; + } + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + delay = 5000000 / priv->rate; + priv->component = &codec->component; + schedule_delayed_work(&priv->dummy_write_work, + usecs_to_jiffies(delay)); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +649,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_DELAYED_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

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[alsa-devel] [PATCH v3 00/15] soundwire: Add a new SoundWire subsystem
by Vinod Koul 30 Nov '17

30 Nov '17

This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v3 v2: https://lkml.org/lkml/2017/11/10/216 v1: https://lkml.org/lkml/2017/10/18/1030 RFC: https://lkml.org/lkml/2016/10/21/395 Changes in v3: - Update the kernel-doc styles and fix included headers for files - handle dev_pm_domain_attach() for defered probe - remove OF placeholders - change regmap license to GPLv2 only Changes in v2: - move documentation into driver-api and do rst conversion - fix documentation comments - add SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) to all source files - rework the transfer logic and paging logic as commented on v1 - remove dummy sysfs fns - registration checks and fixes - remove slave check for regamp as that turned superfluous - remove depends SoundWire symbol - make modalias api arg const - use bitmap for tracking assigned - add counter for report present tracking todo: add the dt-bindings Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (11): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence library soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module MAINTAINERS: Add SoundWire entry Documentation/driver-api/index.rst | 1 + Documentation/driver-api/soundwire/index.rst | 15 + Documentation/driver-api/soundwire/summary.rst | 205 +++++ MAINTAINERS | 10 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 104 +++ drivers/soundwire/Kconfig | 37 + drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 1006 ++++++++++++++++++++++++ drivers/soundwire/bus.h | 123 +++ drivers/soundwire/bus_type.c | 241 ++++++ drivers/soundwire/cadence_master.c | 797 +++++++++++++++++++ drivers/soundwire/cadence_master.h | 95 +++ drivers/soundwire/intel.c | 395 ++++++++++ drivers/soundwire/intel.h | 71 ++ drivers/soundwire/intel_init.c | 246 ++++++ drivers/soundwire/mipi_disco.c | 422 ++++++++++ drivers/soundwire/slave.c | 162 ++++ drivers/soundwire/sysfs.c | 390 +++++++++ include/linux/mod_devicetable.h | 6 + include/linux/regmap.h | 37 + include/linux/soundwire/sdw.h | 542 +++++++++++++ include/linux/soundwire/sdw_intel.h | 71 ++ include/linux/soundwire/sdw_registers.h | 242 ++++++ scripts/mod/devicetable-offsets.c | 4 + scripts/mod/file2alias.c | 15 + 29 files changed, 5263 insertions(+) create mode 100644 Documentation/driver-api/soundwire/index.rst create mode 100644 Documentation/driver-api/soundwire/summary.rst create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

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[alsa-devel] [PATCH v2 0/8] Remove use of "gpio-reset" from DT
by Andrew F. Davis 30 Nov '17

30 Nov '17

Hello all, I was working on fixing up the tlv320aic31xx driver when I noticed its gpio reset was not working, it was due to this driver looking for "gpio-reset" instead of the usual "reset-gpio". A quick check shows this mistake is rare and only copied by one other audio CODECs: $ git grep "reset-gpio" | wc -l 630 $ git grep "gpio-reset" | wc -l 6 Luckliy the two effected drivers only use this reset line when power has been cut to the device, so not only were these optional properties but they had no real functional effect anyway. Lets just fix this before is spreads to drivers were it matters. I've also added fixes tags to each patch so it can be individually back-ported to where this bug was introduced if one wanted to. I'm hoping this can sit on next for a while to get the most testing, just in case I'm wrong about this not breaking anything. Thanks, Andrew Changes from v1: - Move old property to "Deprecated properties" section in binding - Keep backwards compatibility for now in driver with warning - Re-order patches to keep series bisectable - Use "reset-gpios" over "reset-gpio" - Drop taken patch - Rebased on v4.15-rc1 Andrew F. Davis (8): ASoC: tlv320aic31xx: Fix typo in DT binding documentation ASoC: tlv320aic3x: Fix typo in DT binding documentation ASoC: tlv320aic31xx: Fix the reset GPIO OF name ASoC: tlv320aic3x: Fix the reset GPIO OF name ARM: dts: am335x-pepper: Fix the audio CODEC's reset pin ARM: dts: imx6: RDU2: Fix the audio CODEC's reset pin ARM: dts: imx: Fix the audio CODEC's reset pin ARM: dts: omap3-n900: Fix the audio CODEC's reset pin Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 9 ++++++++- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 10 +++++++++- arch/arm/boot/dts/am335x-pepper.dts | 2 +- arch/arm/boot/dts/imx6qdl-gw5903.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-zii-rdu2.dtsi | 4 ++-- arch/arm/boot/dts/omap3-n900.dts | 4 ++-- sound/soc/codecs/tlv320aic31xx.c | 11 +++++++++-- sound/soc/codecs/tlv320aic3x.c | 15 +++++++++++---- 8 files changed, 43 insertions(+), 14 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Change the semantics of the enable option
by Tamaki Nishino 30 Nov '17

30 Nov '17

This patch changes the semantics of the enable option for snd-usb-audio in order to allow users to disable a device specified by either or both of the vendor id and the product id. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

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