March 2022 - Alsa-devel - mailman.alsa-project.org

Re: RFC: usb: gadget: u_audio: Notifying gadget that host started playback/capture?
by Pavel Hofman 30 Oct '23

30 Oct '23

Hi, Dne 08. 09. 21 v 10:21 Pavel Hofman napsal(a): > Hi, > > The current audio gadget has no way to inform the gadget side that the > host side has started playback/capture and that gadget-side alsa > processes should be started. > > Playback/capture processes on the host side do not get stuck without the > gadget side consuming/producing data (OUT requests are ignored in > u_audio_iso_complete, IN ones send initial zeros in their req->buf). > > However, playback/capture processes on the gadget side get stuck without > the host side sending playback OUT packets or capture IN requests and > time out with error. If there was a way to inform the gadget side that > playback/capture has started on the host side, the gadget clients could > react accordingly. > I drafted a simple patch for u_audio.c which defines read-only boolean ctl elems "Capture Requested" and "Playback Requested". Their values are set/reset in methods u_audio_start_capture/playback and u_audio_stop_capture/playback, i.e. at changes of respective altsettings from 0 to 1 and back. Every ctl elem value change sends notification via snd_ctl_notify. The principle works OK for capture/playback start/stop on the host, as monitored by alsactl: pi@raspberrypi:~ $ alsactl monitor hw:UAC2Gadget node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #4 (3,0,0,Capture Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE node hw:UAC2Gadget, #3 (3,0,0,Playback Requested,0) VALUE However at enumeration the USB host switches both playback and capture altsettings repeatedly, generating "fake" events from the gadget side POW. The host even sends regular-sized EP-OUT packets filled with zeros during enumeration (tested on linux only for now). Please is there any way to "detect" the enumeration stage to mask out the "fake" playback/capture start/stop events? The attached patch does not apply cleanly to mainline u_audio.c because it's rebased on other patches not submitted yet but it's only a discussion inducer for now. Thanks a lot, Pavel.

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

27 Oct '23

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

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Regression playing 24bit/96kHz audio on USB audio interface between 5.10.94 and 5.15.21
by Ruud van Asseldonk 02 Jan '23

02 Jan '23

Hi, I have a program that configures a Behringer UMC404HD 192k USB audio interface for playback as follows (pseudocode): snd_pcm_hw_params_set_channels(pcm, hwp, 2); snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S24_3LE); snd_pcm_hw_params_set_access(pcm, hwp, SND_PCM_ACCESS_MMAP_INTERLEAVED); snd_pcm_hw_params_set_rate(pcm, hwp, val=96000, dir=0); snd_pcm_hw_params_set_period_size_near(pcm, hwp, val=513, dir=0); // If I confirm at this point with snd_pcm_hw_params_get_period_size, // I get back 513. snd_pcm_hw_params_set_buffer_size_near(pcm, hwp, val=2052); snd_pcm_hw_params(pcm, hwp); // If I confirm with snd_pcm_hw_params_get_buffer_size, // I get back 2052. On Linux 5.10.94 (specifically Arch Linux ARM package linux-rpi-5.10.94-1), this succeeds, and I can play back audio just fine. On 5.15.21 (specifically linux-rpi-5.15.21-3), all the calls succeed apart from the last one, which returns EINVAL. (My actual program is written in Rust and uses a Rust wrapper library for libasound, https://docs.rs/alsa/0.6.0/alsa/, but the calls should be the ones listed.) I confirmed with snd_pcm_hw_params_test_format that the S24_3LE format is supported. On both kernels it is successful. I confirmed with snd_pcm_hw_params_test_rate that 96 kHz is supported. On both kernels it is successful. There are differences though. When I query these: snd_pcm_hw_params_get_period_size_min snd_pcm_hw_params_get_period_size_max snd_pcm_hw_params_get_buffer_size_min snd_pcm_hw_params_get_buffer_size_max Then on 5.10.94 (the good version), I get: period size min/max: 12/32768, buffer size min/max: 24/65536 But on 5.15.21 (the bad version), I get: period size min/max: 13/71332, buffer size min/max: 26/142663 Also, after I set the period size to 513, if I then query the buffer size min/max, on 5.10.94 I get: buffer size min/max: 1026/65536 But on 5.15.21 I get: buffer size min/max: 1027/142663 In lsmod, snd_usb_audio shows up, and from lsusb -v, the UMC404HD appears to be a standard USB audio interface. The lsusb output is identical on both kernels. Please let me know if there is anything I can do to help further diagnose the issue, or if I should report this elsewhere. Kind regards, Ruud van Asseldonk

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

24 Nov '22

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

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[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

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[PATCH 1/2] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Kai Vehmanen 14 Oct '22

14 Oct '22

Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 221afacbc7fd..b98d7975adea 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2477,6 +2480,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..98633d2684de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), base-commit: d7aca8d8f548570f650d9b1dfedf6902f56f2bce -- 2.34.1

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[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

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[PATCH 1/4] ASoC: fsl-asoc-card: add WM8904 support
by Alifer Moraes 21 Jun '22

21 Jun '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> The codec WM8904 can use internal FLL as PLL source. Whenever the PLL source is not an external MCLK, this source must be setup during hw_params callback otherwise the BCLK could be wrongly evaluated. The SND_SOC_BIAS_PREPARE event is raised after the hw_params callback, so there is no need to set again PLL and SYSCLK and actually there's no need at all the set_bias_level function. Also, when esai is used, a dedicated snd_soc_dai_set_tdm_slot call is required. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/fsl/fsl-asoc-card.c | 47 +++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ee945505281..817dbc1ec635 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "imx-audmux.h" #include "../codecs/sgtl5000.h" +#include "../codecs/wm8904.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" @@ -257,6 +258,38 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) } } + if (of_device_is_compatible(dev->of_node, "fsl,imx-audio-wm8904")) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0, 0, 2, + params_physical_width(params)); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->pll_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + return 0; } @@ -651,6 +684,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->card.set_bias_level = NULL; + priv->codec_priv.mclk_id = WM8904_CLK_FLL; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + } } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -900,6 +946,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8904", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, -- 2.25.1

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[PATCH v2 00/14] ASoC: Intel: avs: Topology and path management
by Cezary Rojewski 10 Jun '22

10 Jun '22

A continuation of avs-driver initial series [1]. This chapter covers path management and topology parsing part which was ealier path of the main series. The two patches that represented these two subjects in the initial series, have been split into many to allow for easier review and discussion. AVS topology is split into two major parts: dictionaries - found within ASoC topology manifest - and path templates. Dictionaries job is to reduce the total amount of memory occupied by topology elements. Rather than having every pipeline and module carry its own information, each refers to specific entry in specific dictionary by provided (from topology file) indexes. In consequence, most struct avs_tplg_xxx are made out of pointers. Path templates are similar to path descriptions found in skylake-driver and they describe how given path shall look like in runtime - number of modules and pipelines that shape it and how they are laid out. A single path template is tied either to FE or BE and thus at most to a single, user-visible endpoint when speaking of FE. Path is a software representation of its ADSP firmware equivalent. It's a logical container for pipelines which are themselves containers - this time for modules i.e. processing units. Depending on the number of audio formats supported, each path template may carry one or more descriptions of given path. During runtime, when audio format is known, description matching said format is selected and used when instantiating path on ADSP firmware side through IPCs. Changes v1 -> v2: - fixed 'set but not used' warning in patch 10/14 as mentioned by ikp builder Changes RFC -> v1: - Plenty of rewordings and spelling fixes as requested by Pierre - any sysfs-functionality-related left in RFC dropped, will be part of separate subject in the future - separated parsing of formatted strings into: "Support link_mask formatted string" patch, increasing series size to 14 patches [1]: https://lore.kernel.org/alsa-devel/20220207122108.3780926-1-cezary.rojewski… Cezary Rojewski (14): ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type include/uapi/sound/intel/avs/tokens.h | 126 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 23 + sound/soc/intel/avs/path.c | 1005 ++++++++++++++++ sound/soc/intel/avs/path.h | 72 ++ sound/soc/intel/avs/topology.c | 1600 +++++++++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ 8 files changed, 3024 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h base-commit: bc55cfd5718c7c23e5524582e9fa70b4d10f2433 -- 2.25.1

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[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

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