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[PATCH] ALSA: hda/realtek: Fix support for Dell Precision 3260
by Jaroslav Kysela 22 Mar '23

22 Mar '23

Unfortunately, in commit 5911d78fabbb a wrong codec patch was selected. The model=alc283-dac-wcaps is equivalent to ALC283_FIXUP_CHROME_BOOK not ALC295_FIXUP_CHROME_BOOK. Fixes: 5911d78fabbb ("ALSA: hda/realtek: Improve support for Dell Precision 3260") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f09a1d7c1b18..3a304010bc9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9260,7 +9260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.2

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[PATCH v2 0/3] ASoC: SOF: ipc4/intel: Support for ChainDMA
by Peter Ujfalusi 22 Mar '23

22 Mar '23

Hi, Changes since v1: - Fix missed mutex unlocking in sof_ipc4_widget_free() Reported by Dan Carpenter (thank you!) On a platform when the DSP is in use, we cannot select individual links to use or not use the DSP, it is either all or none. On some audio endpoint, like HDMI/DP, it is preferred to not use any processing in DSP to reduce the latency and to allow bytestream pass-through (DTS, DD, etc) IPC4 introduces a new type of end-to-end connection within the DSP which is using the host DMA and link DMA in a single buffer, working back-to-back, passing the received data without looking at it or trying to understand the format, content. This mode reduces the latency and allows non PCM streams to be sent from userspace. The feature is enabled per PCM bases, signalled in topology. Regards, Peter --- *** BLURB HERE *** Jyri Sarha (2): ASoC: SOF: ipc4: Add macros for chain-dma message bits ASoC: SOF: ipc4/intel: Add support for chained DMA Ranjani Sridharan (1): ASoC: SOF: topology: Set pipeline widget before updating IPC structures include/sound/sof/ipc4/header.h | 29 +++++++ include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/intel/hda-dai-ops.c | 18 ++++- sound/soc/sof/ipc4-pcm.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.h | 2 + sound/soc/sof/topology.c | 48 ++++++------ 7 files changed, 310 insertions(+), 32 deletions(-) -- 2.40.0

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USB-Audio: Add Sony inzone H7/H9 headset default/HiFi usecase
by GitHub pull_request - opened 22 Mar '23

22 Mar '23

alsa-project/alsa-ucm-conf pull request #293 was opened from laurensmiers: The Sony inzone H7/H9 headset has two usb audio outputs and one usb audio input. One pair of audio output and input is the 'chat' interface which is normally connected to a chat client. The remaining audio ouput is the 'game' interface, which is normally connected to the actual game you're playing. The 'game' interface lists as a separate USB subdevice in alsa. This commit adds the HiFi usecase for the Sony headset which is as follows: - One chat device - One audio output - One audio input - One game device - One audio output Tested with pipewire and pulseaudio, verified with pavucontrol, both devices are listed and audio can be played on both. The audio is mixed in the headset itself. Control wise there isn't much, the only thing that is reported in alsamixer is a volume control for the CAPTURE channel. However, changing the volume of this does not seem to affect the actual audio that is recorded, so I don't see a need to change this when enabling the device. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/293 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/293.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 0/3] ASoC: SOF: ipc4/intel: Support for ChainDMA
by Peter Ujfalusi 22 Mar '23

22 Mar '23

Hi, Changes since v1: - Fix missed mutex unlocking in sof_ipc4_widget_free() Reported by Dan Carpenter (thank you!) On a platform when the DSP is in use, we cannot select individual links to use or not use the DSP, it is either all or none. On some audio endpoint, like HDMI/DP, it is preferred to not use any processing in DSP to reduce the latency and to allow bytestream pass-through (DTS, DD, etc) IPC4 introduces a new type of end-to-end connection within the DSP which is using the host DMA and link DMA in a single buffer, working back-to-back, passing the received data without looking at it or trying to understand the format, content. This mode reduces the latency and allows non PCM streams to be sent from userspace. The feature is enabled per PCM bases, signalled in topology. Regards, Peter --- *** BLURB HERE *** Jyri Sarha (2): ASoC: SOF: ipc4: Add macros for chain-dma message bits ASoC: SOF: ipc4/intel: Add support for chained DMA Ranjani Sridharan (1): ASoC: SOF: topology: Set pipeline widget before updating IPC structures include/sound/sof/ipc4/header.h | 29 +++++++ include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/intel/hda-dai-ops.c | 18 ++++- sound/soc/sof/ipc4-pcm.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.c | 122 +++++++++++++++++++++++++++++- sound/soc/sof/ipc4-topology.h | 2 + sound/soc/sof/topology.c | 48 ++++++------ 7 files changed, 310 insertions(+), 32 deletions(-) -- 2.40.0

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conf: add support for Amlogic AIU and AXG cards
by GitHub pull_request - opened 22 Mar '23

22 Mar '23

alsa-project/alsa-lib pull request #308 was opened from chewitt: This submission adds generic card confs for the Amlogic AIU and AXG audio cards. AIU is used with GXBB/GXL/GXM boards and supports HDMI and hinted S/PDIF output. The AXG conf is used with G12A/G12B/SM1 boards and supports HDMI only output. Hinting S/PDIF support does not work with AXG due to the card always presenting three outputs (with internal routing controlling what they are) so the hint always returns true. This is an AIU device with HDMI and S/PDIF hardware: ``` WP2:~ # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=WETEKPLAY2 WETEK-PLAY2, Default Audio Device sysdefault:CARD=WETEKPLAY2 WETEK-PLAY2, Default Audio Device iec958:CARD=WETEKPLAY2,DEV=0 WETEK-PLAY2, IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=WETEKPLAY2,DEV=0 WETEK-PLAY2, HDMI Audio Output WP2:~ # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: WETEKPLAY2 [WETEK-PLAY2], device 0: fe.dai-link-0 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: WETEKPLAY2 [WETEK-PLAY2], device 1: fe.dai-link-1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` This is an AXG device with HDMI only hardware: ``` N2PLUS:~ # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=ODROIDN2 ODROID-N2, Default Audio Device sysdefault:CARD=ODROIDN2 ODROID-N2, Default Audio Device hdmi:CARD=ODROIDN2,DEV=0 ODROID-N2, HDMI Audio Output N2PLUS:~ # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ODROIDN2 [ODROID-N2], device 0: fe.dai-link-0 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ODROIDN2 [ODROID-N2], device 1: fe.dai-link-1 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ODROIDN2 [ODROID-N2], device 2: fe.dai-link-2 (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` The confs have been in use with the Kodi distro LibreELEC for around two years and other distros supporting Amlogic boards on upstream kernels (Armbian, Manjaro, etc.) have borrowed them from our repo too. The prime reason for submitting them is to allow everyone to drop some patches. Although the confs are known-working it would be great if someone with better alsa knowlege than myself could vet them for correctness. ping @jeromebrunet for awareness Request URL : https://github.com/alsa-project/alsa-lib/pull/308 Patch URL : https://github.com/alsa-project/alsa-lib/pull/308.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/8] Fix sound on ASUS Transformers
by Svyatoslav Ryhel 22 Mar '23

22 Mar '23

- add quirk for headset detection used by some T30 devices (ASUS Transformers, LG Optimus 4X HD and Vu); - add RT5631 and MAX9808x machine drivers - update bindings --- Changes from v1 - fm34 dropped for re-work - quirk for headset detection and rt5631 bringup splitted - minor adjustments in binding updates - improvement of rt5631 rate asignment --- David Heidelberg (1): dt-bindings: sound: nvidia,tegra-audio: add RT5631 CODEC Svyatoslav Ryhel (7): dt-bindings: sound: nvidia,tegra-audio-common: add coupled-mic-hp-detect property ASoC: tegra: Support coupled mic-hp detection ARM: tegra: transformers: update sound nodes ASoC: tegra: Support RT5631 by machine driver ARM: tegra: transformers: bind RT5631 sound nodes dt-bindings: sound: nvidia,tegra-audio: add MAX9808x CODEC ASoC: tegra: Support MAX9808x by machine driver .../sound/nvidia,tegra-audio-common.yaml | 4 + .../sound/nvidia,tegra-audio-max9808x.yaml | 90 +++++++++++++ .../sound/nvidia,tegra-audio-rt5631.yaml | 85 ++++++++++++ arch/arm/boot/dts/tegra20-asus-tf101.dts | 7 +- arch/arm/boot/dts/tegra30-asus-tf201.dts | 17 +++ arch/arm/boot/dts/tegra30-asus-tf300t.dts | 5 +- arch/arm/boot/dts/tegra30-asus-tf300tg.dts | 17 +++ arch/arm/boot/dts/tegra30-asus-tf700t.dts | 17 +++ .../dts/tegra30-asus-transformer-common.dtsi | 9 +- sound/soc/tegra/Kconfig | 18 +++ sound/soc/tegra/tegra_asoc_machine.c | 125 +++++++++++++++++- 11 files changed, 380 insertions(+), 14 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml -- 2.37.2

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Re: [PATCH v2 1/7] dt-bindings: misc: Add the Lantiq PEF2466 E1/T1/J1 framer
by Rob Herring 22 Mar '23

22 Mar '23

On Mon, Mar 20, 2023 at 10:46:19AM +0100, Herve Codina via Alsa-devel wrote: > Received: by alsa1.perex.cz (Postfix, from userid 50401) id 16494F8027B; > Mon, 20 Mar 2023 10:46:37 +0100 (CET) > X-Spam-Checker-Version: SpamAssassin 3.4.6 (2021-04-09) on alsa1.perex.cz > X-Spam-Level: > X-Spam-Status: No, score=-5.2 required=5.0 tests=DKIM_SIGNED,DKIM_VALID, > DKIM_VALID_AU,DKIM_VALID_EF,RCVD_IN_DNSWL_HI,SPF_HELO_NONE,SPF_PASS, > URIBL_BLOCKED shortcircuit=no autolearn=ham autolearn_force=no > version=3.4.6 > Received: from relay6-d.mail.gandi.net (relay6-d.mail.gandi.net > [217.70.183.198]) (using TLSv1.2 with cipher ADH-AES256-GCM-SHA384 > (256/256 bits)) (No client certificate requested) by alsa1.perex.cz > (Postfix) with ESMTPS id 3FF5FF80105 for <alsa-devel(a)alsa-project.org>; > Mon, 20 Mar 2023 10:46:22 +0100 (CET) > DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 3FF5FF80105 > Authentication-Results: alsa1.perex.cz; dkim=pass (2048-bit key, > unprotected) header.d=bootlin.com header.i=(a)bootlin.com > header.a=rsa-sha256 header.s=gm1 header.b=m4O7nLC1 > Received: (Authenticated sender: herve.codina(a)bootlin.com) by > mail.gandi.net (Postfix) with ESMTPSA id 40453C0009; Mon, 20 Mar 2023 > 09:46:20 +0000 (UTC) > DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=bootlin.com; s=gm1; > t=1679305582; > h=from:from:reply-to:subject:subject:date:date:message-id:message-id: > to:to:cc:cc:mime-version:mime-version:content-type:content-type: > content-transfer-encoding:content-transfer-encoding: > in-reply-to:in-reply-to:references:references; > bh=Ieu9Fv38se4lD4z/BVXUHLrVJL9Tx5iKWZgvO8X+VoY=; > b=m4O7nLC1LPZDOI5eM/hmgqouxdkin2veA6CvJhT9kU9rGQALB3ya2fuybMfDvrkTqqBjEd > j6DAxXMgOKgwuUfEsZsp3BFJpoii00hSaf0r2uIbnnGcUrDGVQqUQVEqv51O6VBqnrViQk > PstlJM0lcE9R/AFASd5D/HQGoYYyRY+NKT7xt8g1Ax23Yk/tUG59LXku/skn/4faSLodnU > vV2ng3VMUcoLuvSMJtdYY3hrXEWqUrW1ZogxAFHJNiKuyOELmqZGmNo4B4yAFOEcqqyano > /f4m/7BtT7X1wwPvGu29gg+0aOFrGQq5kb4UNrMoriSQyKnxPRha8zL3J2Jckw== > Date: Mon, 20 Mar 2023 10:46:19 +0100 > From: Herve Codina <herve.codina(a)bootlin.com> > To: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> > Subject: Re: [PATCH v2 1/7] dt-bindings: misc: Add the Lantiq PEF2466 > E1/T1/J1 framer > Message-ID: <20230320104619.468a304b(a)bootlin.com> > In-Reply-To: <96b01241-d57d-a460-4a8b-9e83eaab24ae(a)linaro.org> > References: <20230316122741.577663-1-herve.codina(a)bootlin.com> > <20230316122741.577663-2-herve.codina(a)bootlin.com> > <96b01241-d57d-a460-4a8b-9e83eaab24ae(a)linaro.org> > Organization: Bootlin > X-Mailer: Claws Mail 4.1.1 (GTK 3.24.37; x86_64-redhat-linux-gnu) > MIME-Version: 1.0 > Content-Type: text/plain; charset=UTF-8 > Content-Transfer-Encoding: quoted-printable > Message-ID-Hash: AJZF4VHU24ASVVBCPRMLJCDG4ZDX55LB > X-Message-ID-Hash: AJZF4VHU24ASVVBCPRMLJCDG4ZDX55LB > X-MailFrom: herve.codina(a)bootlin.com > X-Mailman-Rule-Misses: dmarc-mitigation; no-senders; approved; emergency; > loop; banned-address; member-moderation; > header-match-alsa-devel.alsa-project.org-0; > header-match-alsa-devel.alsa-project.org-1; nonmember-moderation; > administrivia; implicit-dest; max-recipients; max-size; news-moderation; > no-subject; digests; suspicious-header > CC: Rob Herring <robh+dt(a)kernel.org>, Krzysztof Kozlowski > <krzysztof.kozlowski+dt(a)linaro.org>, Liam Girdwood <lgirdwood(a)gmail.com>, > Mark Brown <broonie(a)kernel.org>, Derek Kiernan <derek.kiernan(a)xilinx.com>, > Dragan Cvetic <dragan.cvetic(a)xilinx.com>, Arnd Bergmann <arnd(a)arndb.de>, > Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>, Takashi Iwai > <tiwai(a)suse.com>, linux-kernel(a)vger.kernel.org, > devicetree(a)vger.kernel.org, alsa-devel(a)alsa-project.org, Christophe Leroy > <christophe.leroy(a)csgroup.eu>, Thomas Petazzoni > <thomas.petazzoni(a)bootlin.com> > X-Mailman-Version: 3.3.8 > Precedence: list > List-Id: "Alsa-devel mailing list for ALSA developers - > http://www.alsa-project.org" <alsa-devel.alsa-project.org> > Archived-At: <https://mailman.alsa-project.org/hyperkitty/list/alsa-devel@alsa-project.or…> > List-Archive: <https://mailman.alsa-project.org/hyperkitty/list/alsa-devel@alsa-project.or…> > List-Help: <mailto:alsa-devel-request@alsa-project.org?subject=help> > List-Owner: <mailto:alsa-devel-owner@alsa-project.org> > List-Post: <mailto:alsa-devel@alsa-project.org> > List-Subscribe: <mailto:alsa-devel-join@alsa-project.org> > List-Unsubscribe: <mailto:alsa-devel-leave@alsa-project.org> The alsa-devel list doesn't seem to like your emails. The archives (lore) has 2 copies with the 2nd having the original headers in the body. I'm seeing this recently on other senders too. Best I can tell is you sent this as quoted-printable. Rob

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Unlikely big volume range volume range on EMEET webcam
by Pavel Hofman 22 Mar '23

22 Mar '23

Hi, Speaker-webcam eMeet C980 Pro does not play, outputs only silent noise. The digital volume seems to be down. dmesg: [22406.805176] input: HD Webcam eMeet C980 Pro: HD We as /devices/pci0000:00/0000:00:01.2/0000:02:00.0/usb1/1-7/1-7.1/1-7.1:1.0/input/input27 [22406.829048] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22406.933044] usb 1-7.1: Warning! Unlikely big volume range (=10240), cval->res is probably wrong. [22406.933048] usb 1-7.1: [5] FU [PCM Playback Volume] ch = 1, val = -7168/3072/1 [22407.001043] usb 1-7.1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong. [22407.001047] usb 1-7.1: [6] FU [Headset Capture Volume] ch = 1, val = -1536/1536/1 amixer contents: ..... numid=3,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=10240,step=0 : values=10240 | dBminmax-min=-28.00dB,max=12.00dB numid=5,iface=MIXER,name='Headset Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=6,iface=MIXER,name='Headset Capture Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=3072,step=0 : values=3072 | dBminmax-min=-6.00dB,max=6.00dB ..... I tried playing with csetting numid=4 to various values, no effect. Please what should be the correct min,max,res in volume_control_quirks for the 'PCM Playback Volume' control for such device? Capture seems to be OK (maybe the range should be adjusted too for the complete range of control) Also, the device does not seem to get the current rate correctly, but IIUC that should not have any adverse effect (not marked with QUIRK_FLAG_GET_SAMPLE_RATE yet): [22407.178045] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22407.199046] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22407.220045] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22407.282044] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22811.235851] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 [22829.335494] usb 1-7.1: current rate 16000 is different from the runtime rate 48000 cat /proc/asound/card2/stream0 SHENZHEN EMEET TECHNOLOGY CO., LTD. HD Webcam eMeet C980 Pro at usb-0000:02:00. : USB Audio Playback: Status: Stop Interface 3 Altset 1 Format: S16_LE Channels: 1 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 48000 Data packet interval: 1000 us Bits: 16 Channel map: FL Capture: Status: Stop Interface 4 Altset 1 Format: S16_LE Channels: 1 Endpoint: 0x83 (3 IN) (SYNC) Rates: 16000 Data packet interval: 1000 us Bits: 16 alsa-info dump: http://alsa-project.org/db/?f=7af42635d88e858818932578aa8445810916a947 lsusb -v -d 328f:1912 attached. Thank you very much for any hints. Best regards, Pavel.

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Re: [PATCH 1/4] ASoC: es8316: Enable support for S32 LE format and MCLK div by 2
by 沈一超 22 Mar '23

22 Mar '23

> > > This introduces a DT property but there's no documentation for it, but I > > > don't see why we'd want this in the bindings - the driver should be able > > > to tell from the input clock rate and required output/internal clocks if > > > it needs to divide MCLK. > > > The problem here is that I have no knowledge what is the maximum MCLK > > that the codec accepts. According to the datasheet the maximum supported > > frequency of MCLK is 51.2 Mhz. But this doesn't seem to be the case in > > practice since a MCLK of 48Mhz causes noises in the sound output. > > The idea to divide the MCLK by 2 was proposed by a Everest Semiconductor > > engineer. > > So I don't know how to make this generic enough to be activated from the > > codec driver. > > The usual constraint would be that MCLK can be at most some multiple of > LRCLK or something similar (are all the other dividers in the chip set > sensibly for the full scale MCLK?). In any case you're clearly aware of > a specific case where it needs to be divided down which can be > identified even if you're concerned about dividing down for other cases. In practice no dt property is needed. We support 48M mclk, based on enabling MCLK divided by 2 for mclk greater than 24M . You may want code like this: /* enabling mclk divided by 2 for mclk > 24M */ if (es8316->sysclk > 24000000) mclk_div_option = true;

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[PATCH] ASoC: simple-card: add comment to indicate don't remove platforms
by Kuninori Morimoto 22 Mar '23

22 Mar '23

Basically CPU and Platform are different Component, but if CPU is using soc-generic-dmaengine-pcm, same dev will be shared between CPU and Platform, and Simple Card had been supporting it. When we focus to clean up Simple Card driver, we tend to remove platforms if no Platform was selected, but it is wrong because of above reasons. This patch adds comment why we shouldn't remove platforms. In case of CPU is not using soc-generic-dmaengine-pcm, CPU and Platform will be duplicated, but it will be ignored by snd_soc_rtd_add_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 13 ++++++++++++- sound/soc/generic/audio-graph-card2.c | 17 +++++++++++++++++ sound/soc/generic/simple-card.c | 19 ++++++++++++++++++- 3 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index d788f5f23a8a..4e85536a1b26 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -613,10 +613,16 @@ static int graph_count_noml(struct asoc_simple_priv *priv, return -EINVAL; } + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = 1; - li->num[li->link].codecs = 1; li->num[li->link].platforms = 1; + li->num[li->link].codecs = 1; + li->link += 1; /* 1xCPU-Codec */ dev_dbg(dev, "Count As Normal\n"); @@ -637,6 +643,11 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, } if (li->cpu) { + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = 1; li->num[li->link].platforms = 1; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 259544f64df9..d145b74d41e4 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1046,8 +1046,14 @@ static int graph_count_normal(struct asoc_simple_priv *priv, * => lnk: port { endpoint { .. }; }; * }; */ + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(cpu_port); + li->num[li->link].codecs = graph_counter(codec_port); of_node_put(cpu_ep); @@ -1079,6 +1085,11 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, */ if (asoc_graph_is_ports0(lnk)) { + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = graph_counter(rport); /* FE */ li->num[li->link].platforms = graph_counter(rport); } else { @@ -1113,8 +1124,14 @@ static int graph_count_c2c(struct asoc_simple_priv *priv, * }; * }; */ + /* + * DON'T REMOVE platforms + * see + * simple-card.c :: simple_count_noml() + */ li->num[li->link].cpus = li->num[li->link].platforms = graph_counter(codec0); + li->num[li->link].codecs = graph_counter(codec1); of_node_put(ports); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index e98932c16754..fc044978e057 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -509,10 +509,22 @@ static int simple_count_noml(struct asoc_simple_priv *priv, return -EINVAL; } + /* + * DON'T REMOVE platforms + * + * Some CPU might be using soc-generic-dmaengine-pcm. This means CPU and Platform + * are different Component, but are sharing same component->dev. + * Simple Card had been supported it without special Platform selection. + * We need platforms here. + * + * In case of no Platform, it will be Platform == CPU, but Platform will be + * ignored by snd_soc_rtd_add_component(). + */ li->num[li->link].cpus = 1; - li->num[li->link].codecs = 1; li->num[li->link].platforms = 1; + li->num[li->link].codecs = 1; + li->link += 1; return 0; @@ -531,6 +543,11 @@ static int simple_count_dpcm(struct asoc_simple_priv *priv, } if (li->cpu) { + /* + * DON'T REMOVE platforms + * see + * simple_count_noml() + */ li->num[li->link].cpus = 1; li->num[li->link].platforms = 1; -- 2.25.1

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