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[PATCH v5] ASoC: dt-bindings: ak4458: Convert to dtschema
by Saalim Quadri 06 Apr '23

06 Apr '23

Convert the AK4458 audio DAC bindings to DT schema. Signed-off-by: Saalim Quadri <danascape(a)gmail.com> --- Changes: V1 -> V2: Use the correct way for dsd-path property Drop ak4458 label form example V2 -> V3: ak4458 is the only one that does not support dsd-path, so we do not require to define an array V3 -> V4: Add back dsd-path property description V4 -> V5: Fix yaml format as per example-schema .../devicetree/bindings/sound/ak4458.txt | 28 ------- .../bindings/sound/asahi-kasei,ak4458.yaml | 73 +++++++++++++++++++ 2 files changed, 73 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4458.txt create mode 100644 Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt deleted file mode 100644 index 0416c14895d6..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ /dev/null @@ -1,28 +0,0 @@ -AK4458 audio DAC - -This device supports I2C mode. - -Required properties: - -- compatible : "asahi-kasei,ak4458" or "asahi-kasei,ak4497" -- reg : The I2C address of the device for I2C - -Optional properties: -- reset-gpios: A GPIO specifier for the power down & reset pin -- mute-gpios: A GPIO specifier for the soft mute pin -- AVDD-supply: Analog power supply -- DVDD-supply: Digital power supply -- dsd-path: Select DSD input pins for ak4497 - 0: select #16, #17, #19 pins - 1: select #3, #4, #5 pins - -Example: - -&i2c { - ak4458: dac@10 { - compatible = "asahi-kasei,ak4458"; - reg = <0x10>; - reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW> - mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH> - }; -}; diff --git a/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml new file mode 100644 index 000000000000..0db0c8e923a0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/asahi-kasei,ak4458.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/asahi-kasei,ak4458.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4458 audio DAC + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4458 + - asahi-kasei,ak4497 + + reg: + maxItems: 1 + + avdd-supply: + description: Analog power supply + + dvdd-supply: + description: Digital power supply + + reset-gpios: + maxItems: 1 + + mute-gpios: + maxItems: 1 + description: + GPIO used to mute all the outputs + + dsd-path: + description: Select DSD input pins for ak4497 + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: select #16, #17, #19 pins + - const: 1 + description: select #3, #4, #5 pins + +required: + - compatible + - reg + +allOf: + - if: + properties: + compatible: + contains: + const: asahi-kasei,ak4458 + + then: + properties: + dsd-path: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@10 { + compatible = "asahi-kasei,ak4458"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH>; + }; + }; -- 2.34.1

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[PATCH] soundwire: amd: fix an IS_ERR() vs NULL bug
by Dan Carpenter 06 Apr '23

06 Apr '23

The devm_ioremap() function returns NULL on error. It never returns error pointers. Update the error checking accordingly. Fixes: a673a8dfc214 ("soundwire: amd: Add support for AMD Manager driver") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- drivers/soundwire/amd_manager.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 9fb7f91ca182..9bb8ae8c5f32 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -910,9 +910,9 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -ENOMEM; amd_manager->acp_mmio = devm_ioremap(dev, res->start, resource_size(res)); - if (IS_ERR(amd_manager->mmio)) { + if (!amd_manager->mmio) { dev_err(dev, "mmio not found\n"); - return PTR_ERR(amd_manager->mmio); + return -ENOMEM; } amd_manager->instance = pdata->instance; amd_manager->mmio = amd_manager->acp_mmio + -- 2.39.1

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[PATCH] ASoC: amd: yc: Add ASUS M3402RA into DMI table
by Enrico Belleri 06 Apr '23

06 Apr '23

From: Enrico Belleri <idesmi(a)protonmail.com> Fix builtin microphone on ASUS Vivobook S 14 OLED 2022 (M3402RA) Same issue with this model as apparently with other Rembrandt laptops: https://bugzilla.kernel.org/show_bug.cgi?id=216270 Signed-off-by: Enrico Belleri <kilgore.trout(a)idesmi.eu> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 4a69ce702360..80dfb6036859 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M3402RA"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.40.0

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[PATCH] docs: sound: hda: drop mentions of hda-analyzer
by Oswald Buddenhagen 06 Apr '23

06 Apr '23

It's gone from the repos, presumably because nobody ported it to Python 3. I'm leaving hda-emu in, even though it isn't actually usable with current kernels (but with some minor hacking, the encoding tools work). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Documentation/sound/hd-audio/notes.rst | 24 ++++-------------------- 1 file changed, 4 insertions(+), 20 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index a9e35b1f87bd..34865d3b7512 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -308,9 +308,9 @@ Direct Debugging ---------------- If no model option gives you a better result, and you are a tough guy to fight against evil, try debugging via hitting the raw HD-audio -codec verbs to the device. Some tools are available: hda-emu and -hda-analyzer. The detailed description is found in the sections -below. You'd need to enable hwdep for using these tools. See "Kernel +codec verbs to the device. One tool is available: hda-emu. +The detailed description is found in the section below. +You'd need to enable hwdep for using this tool. See the "Kernel Configuration" section. @@ -339,7 +339,7 @@ When this is enabled, the driver creates hardware-dependent devices (one per each codec), and you have a raw access to the device via these device files. For example, ``hwC0D2`` will be created for the codec slot #2 of the first card (#0). For debug-tools such as -hda-verb and hda-analyzer, the hwdep device has to be enabled. +hda-verb, the hwdep device has to be enabled. Thus, it'd be better to turn this on always. ``CONFIG_SND_HDA_RECONFIG`` is a new option, and this depends on the @@ -819,22 +819,6 @@ See README file in the tarball for more details about hda-verb program. -hda-analyzer ------------- -hda-analyzer provides a graphical interface to access the raw HD-audio -control, based on pyGTK2 binding. It's a more powerful version of -hda-verb. The program gives you an easy-to-use GUI stuff for showing -the widget information and adjusting the amp values, as well as the -proc-compatible output. - -The hda-analyzer: - -* https://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer - -is a part of alsa.git repository in alsa-project.org: - -* git://git.alsa-project.org/alsa.git - Codecgraph ---------- Codecgraph is a utility program to generate a graph and visualizes the -- 2.40.0.152.g15d061e6df

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[PATCH] ASoC: dt-bindings: wm8904: Convert to dtschema
by Saalim Quadri 06 Apr '23

06 Apr '23

Convert the WM8904 audio CODEC bindings to DT schema Signed-off-by: Saalim Quadri <danascape(a)gmail.com> --- .../devicetree/bindings/sound/wlf,wm8904.yaml | 66 +++++++++++++++++++ .../devicetree/bindings/sound/wm8904.txt | 33 ---------- 2 files changed, 66 insertions(+), 33 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8904.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000000..78d76c1c1326 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8904 audio CODEC + +description: | + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +maintainers: + - patches(a)opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - wlf,wm8904 + - wlf,wm8912 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + +required: + - compatible + - reg + - clock-names + - clocks + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt deleted file mode 100644 index 66bf261423b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8904.txt +++ /dev/null @@ -1,33 +0,0 @@ -WM8904 audio CODEC - -This device supports I2C only. - -Required properties: - - compatible: "wlf,wm8904" or "wlf,wm8912" - - reg: the I2C address of the device. - - clock-names: "mclk" - - clocks: reference to - <Documentation/devicetree/bindings/clock/clock-bindings.txt> - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - -Examples: - -codec: wm8904@1a { - compatible = "wlf,wm8904"; - reg = <0x1a>; - clocks = <&pck0>; - clock-names = "mclk"; -}; -- 2.34.1

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[PATCH] soundwire: qcom: Fix enumeration of second device on the bus
by Krzysztof Kozlowski 06 Apr '23

06 Apr '23

Some Soundwire buses (like &swr0 on Qualcomm HDK8450) have two devices, which can be brought from powerdown state one after another. We need to keep enumerating them on each slave attached interrupt, otherwise only first will appear. Cc: <stable(a)vger.kernel.org> Fixes: a6e6581942ca ("soundwire: qcom: add auto enumeration support") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Cc: Patrick Lai <quic_plai(a)quicinc.com> --- drivers/soundwire/qcom.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index c296e0bf897b..1e5077d91f59 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -587,14 +587,9 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) case SWRM_INTERRUPT_STATUS_CHANGE_ENUM_SLAVE_STATUS: dev_dbg_ratelimited(swrm->dev, "SWR new slave attached\n"); swrm->reg_read(swrm, SWRM_MCP_SLV_STATUS, &slave_status); - if (swrm->slave_status == slave_status) { - dev_dbg(swrm->dev, "Slave status not changed %x\n", - slave_status); - } else { - qcom_swrm_get_device_status(swrm); - qcom_swrm_enumerate(&swrm->bus); - sdw_handle_slave_status(&swrm->bus, swrm->status); - } + qcom_swrm_get_device_status(swrm); + qcom_swrm_enumerate(&swrm->bus); + sdw_handle_slave_status(&swrm->bus, swrm->status); break; case SWRM_INTERRUPT_STATUS_MASTER_CLASH_DET: dev_err_ratelimited(swrm->dev, -- 2.34.1

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[PATCH] ALSA: emu10k1: documentation updates
by Oswald Buddenhagen 06 Apr '23

06 Apr '23

- Less misinformation - Language and whitespace fixups Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Documentation/sound/alsa-configuration.rst | 5 ++-- Documentation/sound/cards/audigy-mixer.rst | 27 ++++++++++++--------- Documentation/sound/cards/sb-live-mixer.rst | 17 +++++++------ 3 files changed, 28 insertions(+), 21 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 5f31fa5e2435..af71c68f1e4e 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -723,9 +723,10 @@ Module for EMU10K1/EMU10k2 based PCI sound cards. * Sound Blaster Live! * Sound Blaster PCI 512 -* Emu APS (partially supported) * Sound Blaster Audigy - +* E-MU APS (partially supported) +* E-MU DAS + extin bitmap of available external inputs for FX8010 (see below) extout diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst index c506f8d16f2e..aa176451d5b5 100644 --- a/Documentation/sound/cards/audigy-mixer.rst +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -19,30 +19,30 @@ Digital mixer controls These controls are built using the DSP instructions. They offer extended functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be clipped +(set to maximal or minimal value without checking for overflow). Explanation of used abbreviations: DAC digital to analog converter ADC analog to digital converter I2S one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) + (this standard is used for connecting standalone D/A and A/D converters) LFE - low frequency effects (subwoofer signal) + low frequency effects (used as subwoofer signal) AC97 - a chip containing an analog mixer, DAC and ADC converters + a chip containing an analog mixer, D/A and A/D converters IEC958 S/PDIF FX-bus the EMU10K2 chip has an effect bus containing 64 accumulators. - Each of the synthesizer voices can feed its output to these accumulators - and the DSP microcontroller can operate with the resulting sum. + Each of the synthesizer voices can feed its output to these accumulators + and the DSP microcontroller can operate with the resulting sum. name='PCM Front Playback Volume',index=0 ---------------------------------------- @@ -218,8 +218,8 @@ LFE outputs. name='IEC958 Optical Raw Playback Switch',index=0 ------------------------------------------------- If this switch is on, then the samples for the IEC958 (S/PDIF) digital -output are taken only from the raw FX8010 PCM, otherwise standard front -PCM samples are taken. +output are taken only from the raw iec958 ALSA PCM device (which uses +accumulators 20 and 21 for left and right PCM by default). PCM stream related controls @@ -237,8 +237,8 @@ as follows: name='EMU10K1 PCM Send Routing',index 0-31 ------------------------------------------ -This control specifies the destination - FX-bus accumulators. There 24 -values with this mapping: +This control specifies the destination - FX-bus accumulators. There are 24 +values in this mapping: * 0 - mono, A destination (FX-bus 0-63), default 0 * 1 - mono, B destination (FX-bus 0-63), default 1 @@ -306,6 +306,9 @@ MANUALS/PATENTS ftp://opensource.creative.com/pub/doc ------------------------------------- +Note that the site is defunct, but the documents are available +from various other locations. + LM4545.pdf AC97 Codec diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst index 357fcd619d39..819886634400 100644 --- a/Documentation/sound/cards/sb-live-mixer.rst +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -15,7 +15,7 @@ The ALSA driver programs this portion of chip by default code IEC958 (S/PDIF) raw PCM ======================= -This PCM device (it's the 4th PCM device (index 3!) and first subdevice +This PCM device (it's the 3rd PCM device (index 2!) and first subdevice (index 0) for a given card) allows to forward 48kHz, stereo, 16-bit little endian streams without any modifications to the digital output (coaxial or optical). The universal interface allows the creation of up @@ -33,24 +33,24 @@ Digital mixer controls These controls are built using the DSP instructions. They offer extended functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the -neutral position leaving the signal unchanged. Note that if the same destination -is mentioned in multiple controls, the signal is accumulated and can be wrapped -(set to maximal or minimal value without checking of overflow). +neutral position leaving the signal unchanged. Note that if the same destination +is mentioned in multiple controls, the signal is accumulated and can be clipped +(set to maximal or minimal value without checking for overflow). Explanation of used abbreviations: DAC digital to analog converter ADC analog to digital converter I2S one-way three wire serial bus for digital sound by Philips Semiconductors - (this standard is used for connecting standalone DAC and ADC converters) + (this standard is used for connecting standalone D/A and A/D converters) LFE - low frequency effects (subwoofer signal) + low frequency effects (used as subwoofer signal) AC97 - a chip containing an analog mixer, DAC and ADC converters + a chip containing an analog mixer, D/A and A/D converters IEC958 S/PDIF FX-bus @@ -313,6 +313,9 @@ MANUALS/PATENTS ftp://opensource.creative.com/pub/doc ------------------------------------- +Note that the site is defunct, but the documents are available +from various other locations. + LM4545.pdf AC97 Codec m2049.pdf -- 2.40.0.152.g15d061e6df

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[PATCH] ALSA: emu10k1: update label & help in config system
by Oswald Buddenhagen 06 Apr '23

06 Apr '23

The newer E-MU cards weren't mentioned at all. The "partially supported" is removed ahead of it becoming mostly untrue. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a55836225401..861958451ef5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -461,17 +461,17 @@ config SND_INDIGODJX will be called snd-indigodjx config SND_EMU10K1 - tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" + tristate "Emu10k1 (SB Live!, Audigy, E-MU APS/0404/1010/1212/1616/1820)" select FW_LOADER select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC select SND_TIMER select SND_SEQ_DEVICE if SND_SEQUENCER != n depends on ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, - Audigy and E-mu APS (partially supported) soundcards. + Audigy and E-MU APS/0404/1010/1212/1616/1820 soundcards. The confusing multitude of mixer controls is documented in <file:Documentation/sound/cards/sb-live-mixer.rst> and -- 2.40.0.152.g15d061e6df

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[PATCH] ALSA: emu10k1: don't create old pass-through playback device on Audigy
by Oswald Buddenhagen 06 Apr '23

06 Apr '23

It could have never worked, as snd_emu10k1_fx8010_playback_prepare() and snd_emu10k1_fx8010_playback_hw_free() assume the emu10k1 offset for the ETRAM, and the default DSP code includes no handler for it. It also wouldn't make a lot of sense to make it work, as Audigy has an own, much simpler, pass-through mechanism. So just skip creation of the device. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 48af77ae8020..b1a35b41411a 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1781,17 +1781,21 @@ int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device) struct snd_kcontrol *kctl; int err; - err = snd_pcm_new(emu->card, "emu10k1 efx", device, 8, 1, &pcm); + err = snd_pcm_new(emu->card, "emu10k1 efx", device, emu->audigy ? 0 : 8, 1, &pcm); if (err < 0) return err; pcm->private_data = emu; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_fx8010_playback_ops); + if (!emu->audigy) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_fx8010_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1_capture_efx_ops); pcm->info_flags = 0; - strcpy(pcm->name, "Multichannel Capture/PT Playback"); + if (emu->audigy) + strcpy(pcm->name, "Multichannel Capture"); + else + strcpy(pcm->name, "Multichannel Capture/PT Playback"); emu->pcm_efx = pcm; /* EFX capture - record the "FXBUS2" channels, by default we connect the EXTINs -- 2.40.0.152.g15d061e6df

2 1

[PATCH] ALSA: emu10k1: fix capture interrupt handler unlinking
by Oswald Buddenhagen 06 Apr '23

06 Apr '23

Due to two copy/pastos, closing the MIC or EFX capture device would make a running ADC capture hang due to unsetting its interrupt handler. In principle, this would have also allowed dereferencing dangling pointers, but we're actually rather thorough at disabling and flushing the ints. While it may sound like one, this actually wasn't a hypothetical bug: PortAudio will open a capture stream at startup (and close it right away) even if not asked to. If the first device is busy, it will just proceed with the next one ... thus killing a concurrent capture. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 48af77ae8020..908f76f1bb9f 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1236,7 +1236,7 @@ static int snd_emu10k1_capture_mic_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_mic_interrupt = NULL; emu->pcm_capture_mic_substream = NULL; return 0; } @@ -1344,7 +1344,7 @@ static int snd_emu10k1_capture_efx_close(struct snd_pcm_substream *substream) { struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); - emu->capture_interrupt = NULL; + emu->capture_efx_interrupt = NULL; emu->pcm_capture_efx_substream = NULL; return 0; } -- 2.40.0.152.g15d061e6df

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