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

[alsa-devel] Boundary calculation
by Ricardo Biehl Pasquali 17 Jul '18

17 Jul '18

Hello. I can't get the tricky boundary calculation. Perhaps I'm missing some obvious thing, or maybe it's really some circumvent for a specific case that was not commented. The current calculation for boundary is: boundary = buffer_size; while (boundary * 2 <= LONG_MAX - buffer_size) boundary *= 2; I've checked out some repositories, specially alsa-driver in ALSA git, and I realized this calculation exists at least since 2001. I guess it was Jaroslav who wrote. Visualization of the calculation (B = buffer_size): LONG_MAX ------------------------------- \______________/ \__/ B * 4 B \______/ B * 2 \__/ B Why isn't a simpler calculation used instead? e.g.: /* closest multiple of buffer size less or equal LONG_MAX */ boundary = LONG_MAX - LONG_MAX % buffer_size; LONG_MAX ------------------------------- \__________________________/ B * 6 Cheers! pasquali

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Re: [alsa-devel] [PATCH RFC] usb: add usb_fill_iso_urb()
by Clemens Ladisch 17 Jul '18

17 Jul '18

Sebastian Andrzej Siewior wrote: > sound/ is (almost) the only part where struct usb_iso_packet_descriptor > init does not fit. It looks like it is done just before urb_submit() and > could be avoided there (moved to the init funtcion instead). For playback, the packet lengths change dynamically, because the nominal sample rate (let alone the actual sample rate) often is not an integer multiple of the 1 kHz/8 kHz USB frame rate. Regards, Clemens

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[alsa-devel] [PATCH 0/3] Make the audio component binding more generic
by Takashi Iwai 16 Jul '18

16 Jul '18

Hi, this is a preliminiary patch set to convert the existing i915 / HD-audio component binding to be applicable to other drivers like radeon / amdgpu. This patchset itself doesn't change the functionality but only renames and split to a new drm_audio_component stuff from i915_audio_component. The actual usage of the new API will follow once after this one gets reviewed / accepted. The whole patches (including this patchset) are found in topic/hda-acomp branch of sound.git tree. BTW, since the whole stuff is about the audio binding, I suppose these will go through sound git tree. Let me know if anyone has concerns. Thanks! Takashi === Takashi Iwai (3): drm/i915: Split audio component to a generic type ALSA: hda/i915: Associate audio component with devres ALSA: hda: Make audio component support more generic drivers/gpu/drm/i915/Kconfig | 1 + drivers/gpu/drm/i915/intel_audio.c | 22 +- include/drm/drm_audio_component.h | 138 ++++++++++++ include/drm/i915_component.h | 85 +------- include/sound/hda_component.h | 61 ++++++ include/sound/hda_i915.h | 37 +--- include/sound/hdaudio.h | 6 +- sound/hda/Kconfig | 7 +- sound/hda/Makefile | 1 + sound/hda/hdac_component.c | 332 ++++++++++++++++++++++++++++ sound/hda/hdac_i915.c | 335 ++--------------------------- sound/pci/hda/patch_hdmi.c | 57 +++-- sound/soc/codecs/hdac_hdmi.c | 10 +- 13 files changed, 624 insertions(+), 468 deletions(-) create mode 100644 include/drm/drm_audio_component.h create mode 100644 include/sound/hda_component.h create mode 100644 sound/hda/hdac_component.c -- 2.18.0

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Re: [alsa-devel] 答复: 答复: 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Takashi Iwai 16 Jul '18

16 Jul '18

On Tue, 10 Jul 2018 09:44:42 +0200, Qu, Jim wrote: > > Hi Takashi, > > I am not expert in audio driver, so I just update some test result, please help triage the issue. > > With audio runtime pm: What does this mean? Is it the stock 4.17.x (or 4.18-rc)? Please clarify your test environment at first: which kernel, which hardware setup. > 1. ubuntu audio setting use pactl to query audio card/output sink. Attachment is pactl output with audio runtime pm. > 2. cat /proc/asound/card0/eld* can get nothing. > > Without audio runtime pm: > 1. pactl can get available audio output/sink > 2. can get eld info from eld#0.1 > > IMO, the issue should be: > > Current operations like HDMI hotplug in/out, pactl command, they do > not access audio HW, so the audio could not resume back at this > period. Sorry, not understood. Why they don't access the audio hardware? > Therefore, upper software could not get HDMI ELD info, can select a available card/output sink. > I am also confuse that if there is no ELD info, how driver to steam audio device to wake up itself? It's sort of a chicken-or-egg question. As long as it's i915 and HD-audio, the hotplug and ELD info transfer doesn't trigger the runtime PM since it's done via the direct callback. And ELD value is cached in HD-audio side. The exception is that if the notification is done during the PM operation. But, the connection and ELD query is performed at the end of codec resume, hence it'll be covered there. So in theory, ELD information should be passed from the GPU to HD-audio no matter whether runtime PM is enabled or not. BTW, please stop top-posting. It makes the thread readability awfully worse. thanks, Takashi > > Thanks > JimQu > > ________________________________________ > 发件人: Takashi Iwai <tiwai(a)suse.de> > 发送时间: 2018年7月10日 1:09 > 收件人: Qu, Jim > 抄送: Daniel Vetter; Alex Deucher; alsa-devel(a)alsa-project.org; amd-gfx(a)lists.freedesktop.org; dri-devel(a)lists.freedesktop.org; Deucher, Alexander > 主题: Re: 答复: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > On Mon, 09 Jul 2018 18:05:09 +0200, > Qu, Jim wrote: > > > > Hi Takashi, > > > > Not intel, but it is AMD APU+ AMD GFX, the APU has a local HDMI port for extension. And dGPU is only for offloading render via PRIME. > > > > Originally, the HDA driver before v4.17, there is a bug, that all the audio is set to CLIENT_DIS, so the when the dGPU suspend, the audio which is bound to iGPU will also be suspend. > > > > Now, after Lukas' patches. The audio will auto suspend. It make ubuntu audio setting could not detect HDMI audio, even if HDMI has light up. > > OK, and it has all necessary patches including the one Lukas > suggested, right? > > If so, figure out what you're actually seeing as "PA could not detect > HDMI audio". For example, is it the HDMI (jack) detection (giving > false even if it's connected)? Or is it an error at opening the given > device? Does the driver give the proper ELD bytes as well as the jack > state? > > > Takashi > > > > > Thanks > > JimQu > > > > -----邮件原件----- > > 发件人: Takashi Iwai <tiwai(a)suse.de> > > 发送时间: 2018年7月9日 23:58 > > 收件人: Daniel Vetter <daniel(a)ffwll.ch> > > 抄送: Alex Deucher <alexdeucher(a)gmail.com>; alsa-devel(a)alsa-project.org; amd-gfx(a)lists.freedesktop.org; Qu, Jim <Jim.Qu(a)amd.com>; dri-devel(a)lists.freedesktop.org; Deucher, Alexander <Alexander.Deucher(a)amd.com> > > 主题: Re: [alsa-devel] 答复: [PATCH] vgaswitchroo: set audio client id according to bound gpu client id > > > > On Mon, 09 Jul 2018 17:56:43 +0200, > > Daniel Vetter wrote: > > > > > > On Mon, Jul 09, 2018 at 05:04:22PM +0200, Takashi Iwai wrote: > > > > On Mon, 09 Jul 2018 15:58:51 +0200, > > > > Alex Deucher wrote: > > > > > > > > > > On Mon, Jul 9, 2018 at 6:16 AM, Qu, Jim <Jim.Qu(a)amd.com> wrote: > > > > > > Hi Lukas, > > > > > > > > > > > > Thanks to your explanation, and see comments in line. > > > > > > > > > > > > > > > > > > Do you need to runtime resume the HDA controller even if user > > > > > > space isn't streaming audio? Why, and in which situation exactly? > > > > > > > > > > > > Jim: OEM system uses pactl to queiry audio card and audio output sink, since the audio has power down by runtime pm, so the audio card and output sink are all unavailable. they could not select the available HDMI audio for audio playing. > > > > > > > > > > > > You're saying above that the HDA controller isn't runtime > > > > > > resumed on hotplug of a display. Is that necessary to retrieve ELD or something? > > > > > > I'm not sure if there's code in the HDA driver to acquire a > > > > > > runtime PM ref on HPD, but maybe it's necessary. Or maybe the > > > > > > code is there but somehow no HPD interrupt is received by the HDA driver? > > > > > > > > > > > > Jim: So far, I do not find any code about audio response HPD in kernel. the HPD interrupt will sent to user mode via uevent, not sure whether audio user mode driver can receive the event or not. > > > > > > > > > > On the gfx side at least, we can get a hotplug event via ACPI > > > > > (depending on the OEM design) if displays are attached/detached > > > > > while the dGPU is powered down. I suppose the gfx driver could > > > > > call into the audio driver during one of those events. On the gfx > > > > > side at least we just generate the gfx hotplug event and let userspace deal with it. > > > > > > > > IMO, a more proper way would be to have the direct communication > > > > between the graphics driver and the audio driver like i915 driver > > > > does. Then the audio driver can get plug/unplug event at more > > > > accurate timing without races. > > > > > > > > (Though, it will cause another mess wrt the weak module dependency, > > > > but it's another story :) > > > > > > Just don't do what snd-hda-intel did and instead implement the > > > component stuff properly :-) > > > > A patch is welcome :) > > > > > > thanks, > > > > Takashi > > > > > -Daniel > > > > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Alex > > > > > > > > > > > > > > > > > Thanks > > > > > > JimQu > > > > > > > > > > > > ________________________________________ > > > > > > 发件人: Lukas Wunner <lukas(a)wunner.de> > > > > > > 发送时间: 2018年7月9日 17:27 > > > > > > 收件人: Qu, Jim > > > > > > 抄送: alsa-devel(a)alsa-project.org; > > > > > > dri-devel(a)lists.freedesktop.org; Deucher, Alexander; > > > > > > amd-gfx(a)lists.freedesktop.org > > > > > > 主题: Re: [PATCH] vgaswitchroo: set audio client id according to > > > > > > bound gpu client id > > > > > > > > > > > > On Mon, Jul 09, 2018 at 08:52:33AM +0000, Qu, Jim wrote: > > > > > >> Now, I found the audio device will auto suspend even if the gpu > > > > > >> is active, and if I plug in a HDMI device it also do not resume back. > > > > > >> > > > > > >> 1. Did you encounter similar issue before? > > > > > >> 2. audio will auto suspend as default at beginning of boot, is > > > > > >> it expect behaviour? > > > > > > > > > > > > Yes, that's expected. Once you start streaming audio to > > > > > > attached displays, user space opens the codec device and this > > > > > > causes the HDA controller to runtime resume. If the discrete > > > > > > GPU is suspended at that moment, it will be powered on and kept powered on as long as user space is streaming audio. > > > > > > > > > > > > Do you need to runtime resume the HDA controller even if user > > > > > > space isn't streaming audio? Why, and in which situation exactly? > > > > > > > > > > > > You're saying above that the HDA controller isn't runtime > > > > > > resumed on hotplug of a display. Is that necessary to retrieve ELD or something? > > > > > > I'm not sure if there's code in the HDA driver to acquire a > > > > > > runtime PM ref on HPD, but maybe it's necessary. Or maybe the > > > > > > code is there but somehow no HPD interrupt is received by the HDA driver? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Lukas > > > > > > _______________________________________________ > > > > > > dri-devel mailing list > > > > > > dri-devel(a)lists.freedesktop.org > > > > > > https://lists.freedesktop.org/mailman/listinfo/dri-devel > > > > > _______________________________________________ > > > > > Alsa-devel mailing list > > > > > Alsa-devel(a)alsa-project.org > > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > _______________________________________________ > > > > dri-devel mailing list > > > > dri-devel(a)lists.freedesktop.org > > > > https://lists.freedesktop.org/mailman/listinfo/dri-devel > > > > > > -- > > > Daniel Vetter > > > Software Engineer, Intel Corporation > > > http://blog.ffwll.ch > > > > [2 pactl_with_runtimepm <application/octet-stream (base64)>] >

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[alsa-devel] [V2 1/2] ALSA: HDA: use PCI_BASE_CLASS_DISPLAY to replace PCI_CLASS_DISPLAY_VGA
by Jim Qu 16 Jul '18

16 Jul '18

Except PCI_CLASS_DISPLAY_VGA, some PCI class is sometimes PCI_CLASS_DISPLAY_3D or PCI_CLASS_DISPLAY_OTHER. Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850..14733ff 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1429,7 +1429,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk
by yokota.hgml＠gmail.com 16 Jul '18

16 Jul '18

This adds some required quirk when uses headset or headphone on Panasonic CF-SZ6. Signed-off-by: YOKOTA Hiroshi <yokota.hgml(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+)

2 1

[alsa-devel] [PATCH] ALSA: hda: add mute led support for HP ProBook 455 G5
by Po-Hsu Lin 16 Jul '18

16 Jul '18

Audio mute led does not work on HP ProBook 455 G5, this can be fixed by using CXT_FIXUP_MUTE_LED_GPIO to support it. BugLink: https://bugs.launchpad.net/bugs/1781763 Reported-by: James Buren Signed-off-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e7fcfc3..f641c20 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -964,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.7.4

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[alsa-devel] [PATCH 1/2] ALSA: HDA: use PCI_BASE_CLASS_DISPLAY to cover more display adapters
by Jim Qu 16 Jul '18

16 Jul '18

Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850..e0064567 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -31,7 +31,7 @@ * CHANGES: * * 2004.12.01 Major rewrite by tiwai, merged the work of pshou - * + * */ #include <linux/delay.h> @@ -390,8 +390,8 @@ static char *driver_short_names[] = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", - [AZX_DRIVER_TERA] = "HDA Teradici", - [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", @@ -1429,7 +1429,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), pci->bus->number, 0); if (p) { - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) return p; pci_dev_put(p); } -- 2.7.4

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[alsa-devel] [PATCH 0/5] UAC3: Add Selectors and Processing Units.
by Jorge Sanjuan 16 Jul '18

16 Jul '18

This patchset is motivated by the addition of Multi Function Processing Units (MFPU) to an UAC3 topology where there could be signal processing algorithims applied to the audio signal. The MFPUs themself don't provide any useful control, they offer a description of what algorithims it supports and it is for Selector Units to take the bypass control of the dry (unprocessed) signal and the modified one. Moreover, Up/Down mixers may be needed in this topologies to control which outputs of the logical output cluster from the MFPU are to be passed as the final modified audio signal. These patches add support for Selector Units and Processing units for UAC3 and adds a couple fixes that I found while implemeting them: 1) Defualt naming of the virtual terminals was not accurate due to codes overlap between the three UAC standards. 2) UAC2 parsing of processing units was using UAC1 controls bitmap. Based on: next-20180711 Jorge Sanjuan (5): ALSA: usb-audio: Add support for Selector Units in UAC3 ALSA: usb-audio: Processing Unit controls parsing in UAC2 ALSA: usb-audio: Add support for Processing Units in UAC3 ALSA: usb-audio: Unify virtual type units type to UAC3 values ALSA: usb-audio: Tidy up logic for Processing Unit min/max values include/linux/usb/audio-v3.h | 15 +++ include/uapi/linux/usb/audio.h | 49 ++++++++-- sound/usb/mixer.c | 212 +++++++++++++++++++++++++++++++++-------- 3 files changed, 225 insertions(+), 51 deletions(-) -- 2.11.0

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[alsa-devel] Applied "ASoC: hdmi-codec: fix routing" to the asoc tree
by Mark Brown 16 Jul '18

16 Jul '18

The patch ASoC: hdmi-codec: fix routing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d30e23d69981a4b665f5ce8711335df986576389 Mon Sep 17 00:00:00 2001 From: Russell King <rmk+kernel(a)armlinux.org.uk> Date: Sat, 14 Jul 2018 16:01:06 +0100 Subject: [PATCH] ASoC: hdmi-codec: fix routing Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.18.0.rc2

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[alsa-devel] [PATCH v4] ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver
by Naveen Manohar 16 Jul '18

16 Jul '18

Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v1->v2: reverse xmas tree alignment of var declaration, Add Acked-by. v2->v3: corrected: platform-name for echo-ref dai & change-log position. v3->v4: turn-off sys_clk instead of PLL, when use-case is not ran to save power sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 643 ++++++++++++++++++++++++++ 3 files changed, 659 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2479748..cccda87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507..87ef8b4 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 0000000..c4b94e2 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); -- 2.7.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: AMD: Send correct channel for configuring DMA descriptors
by Akshu Agrawal 16 Jul '18

16 Jul '18

Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for ACP-I2S transfer. With recent patches ch1 is used to define channel order number 1 and ch2 as channel order number 2. Thus, Playback: ch1:SYSMEM->ACP ch2:ACP->I2S Capture: ch1:I2S->ACP ch1:ACP->SYSMEM Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 65c1033..eeb8677 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -322,17 +322,27 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } @@ -995,16 +1005,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; -- 1.9.1

2 3

[alsa-devel] [PATCH v5 0/5] ASoC: add driver for Atmel I2S controller
by Codrin Ciubotariu 16 Jul '18

16 Jul '18

This is a rework of Cyrille's patches named: [PATCH v3 0/2] ASoC: add driver for Atmel I2S controller https://lkml.org/lkml/2015/9/29/454 This is the version 5 of the series, and addresses the received feedback on the mailing lists. This series applies on top of clk-next branch of clk/linux.git. Based on received feedback, I created a mux clock driver to select the I2S clock source, that also includes proper devicetree bindings and nodes. Also, I added the I2S nodes in sama5d2's devicetree, with the missing pin muxing for the second I2S controller. Alexandre Belloni plans to change the at91 fine grained clock nodes and he recommended to leave the bindings for this clock driver in the same (wrong) manner, so he could fix them all at once. This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v4 -> v5 - removed patches: [PATCH v4 4/7] ASoC: atmel-i2s: dt-bindings: add DT bindings for I2S controller [PATCH v4 5/7] ASoC: atmel-i2s: add driver for the new Atmel I2S controller from this series since they were applied to the asoc tree at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git - addressed comments received on the mux clock; v3 -> v4 - as suggested by Rob Herring: - added a clock mux driver for I2S's clock control bit; - more precise description of I2S's devicetree bindings; - removed SoC and internal detalls from bindings; - addressed comments from Mark Brown; - added devicetree nodes and pin muxing for I2S; v2 -> v3 - fix the coding style, add some more comments and add a section dedicated to sama5d2 SoCs in the DT binding documentation as suggested by Mark Brown. v1 -> v2 - initialize dev->dev before calling dev->caps->mck_init() Codrin Ciubotariu (3): dt-bindings: clk: at91: add an I2S mux clock clk: at91: add I2S clock mux driver ARM: dts: at91: sama5d2: add I2S clock muxing nodes Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add nodes for I2S controllers ARM: dts: at91: sama5d2 Xplained: add pin muxing for I2S .../devicetree/bindings/clock/at91-clock.txt | 35 ++++++ arch/arm/boot/dts/at91-sama5d2_xplained.dts | 28 +++++ arch/arm/boot/dts/sama5d2.dtsi | 52 ++++++++ arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 1 + drivers/clk/at91/clk-i2s-mux.c | 116 ++++++++++++++++++ 6 files changed, 236 insertions(+) create mode 100644 drivers/clk/at91/clk-i2s-mux.c -- 2.17.1

4 13

[alsa-devel] [PATCH] ALSA: cs46xx: remove redundant pointer 'ins'
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer 'ins' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'ins' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/cs46xx/dsp_spos_scb_lib.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index abb01ce66983..8d0a3d357345 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -73,13 +73,10 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, { struct proc_scb_info * scb_info = entry->private_data; struct dsp_scb_descriptor * scb = scb_info->scb_desc; - struct dsp_spos_instance * ins; struct snd_cs46xx *chip = scb_info->chip; int j,col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - ins = chip->dsp_spos_instance; - mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: ali5451: remove redundant pointer 'codec'
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer 'codec' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'codec' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ali5451/ali5451.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 39547e32e584..9f569379b77e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1484,12 +1484,9 @@ static struct snd_pcm_hardware snd_ali_capture = static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime) { struct snd_ali_voice *pvoice = runtime->private_data; - struct snd_ali *codec; - if (pvoice) { - codec = pvoice->codec; + if (pvoice) snd_ali_free_voice(pvoice->codec, pvoice); - } } static int snd_ali_open(struct snd_pcm_substream *substream, int rec, -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: sb8: remove redundant pointer runtime
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer runtime is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/sb/sb8_main.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index d45df5c54423..80e7dcaa551f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -381,7 +381,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) { struct snd_pcm_substream *substream; - struct snd_pcm_runtime *runtime; snd_sb_ack_8bit(chip); switch (chip->mode) { @@ -391,7 +390,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; - runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); @@ -402,7 +400,6 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; - runtime = substream->runtime; if (chip->capture_format == SB_DSP_INPUT) snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: gus: remove redundant pointer private_data
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer private_data is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'private_data' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/gus/gus_reset.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 3d1fed0c2620..59b3f683d49b 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -292,7 +292,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) { unsigned long flags; void (*private_free)(struct snd_gus_voice *voice); - void *private_data; if (voice == NULL || !voice->use) return; @@ -300,7 +299,6 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) snd_gf1_clear_voices(gus, voice->number, voice->number); spin_lock_irqsave(&gus->voice_alloc, flags); private_free = voice->private_free; - private_data = voice->private_data; voice->private_free = NULL; voice->private_data = NULL; if (voice->pcm) -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: es1688: remove redundant pointer chip
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Pointer chip is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'chip' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/es1688/es1688.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..3dfe7e592c25 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -260,7 +260,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, struct snd_card *card; static unsigned int dev; int error; - struct snd_es1688 *chip; if (snd_es968_pnp_is_probed) return -EBUSY; @@ -276,7 +275,6 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, sizeof(struct snd_es1688), &card); if (error < 0) return error; - chip = card->private_data; error = snd_card_es968_pnp(card, dev, pcard, pid); if (error < 0) { -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: opl3: remove redundant pointer opl3
by Colin King 16 Jul '18

16 Jul '18

From: Colin Ian King <colin.king(a)canonical.com> Variable opl3 is being assigned but is never used hence it is redundant and can be removed. Cleans up several clang warnings: warning: variable 'opl3' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/drivers/opl3/opl3_midi.c | 12 ------------ sound/drivers/opl3/opl3_oss.c | 6 ------ 2 files changed, 18 deletions(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index bb3f3a5a6951..71cd5a2fbe82 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -721,9 +721,6 @@ void snd_opl3_note_off(void *p, int note, int vel, */ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -735,9 +732,6 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha */ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -861,9 +855,6 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); @@ -876,9 +867,6 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset) { - struct snd_opl3 *opl3; - - opl3 = p; #ifdef DEBUG_MIDI snd_printk(KERN_DEBUG "SYSEX\n"); #endif diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 22c3e4bca220..8a0ce3f43f42 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -233,11 +233,8 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: snd_printk(KERN_ERR "OPL3: " @@ -261,11 +258,8 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, /* reset device */ static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg) { - struct snd_opl3 *opl3; - if (snd_BUG_ON(!arg)) return -ENXIO; - opl3 = arg->private_data; return 0; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: remove clock divider and pll setup from zylonite and magician
by Daniel Mack 16 Jul '18

16 Jul '18

The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/magician.c | 105 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 1 insertion(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..2746725a898c 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -201,23 +115,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- 2.17.1

2 2

[alsa-devel] WARNING in __snd_rawmidi_transmit_ack
by syzbot 16 Jul '18

16 Jul '18

Hello, syzbot found the following crash on: HEAD commit: 37b5dca2898d Merge tag 'rtc-4.18-2' of git://git.kernel.or.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=12204d62400000 kernel config: https://syzkaller.appspot.com/x/.config?x=25856fac4e580aa7 dashboard link: https://syzkaller.appspot.com/bug?extid=52f83f0ea8df16932f7f compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?x=148ea9a4400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+52f83f0ea8df16932f7f(a)syzkaller.appspotmail.com random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: cc1: uninitialized urandom read (8 bytes read) IPVS: ftp: loaded support on port[0] = 21 WARNING: CPU: 0 PID: 4608 at sound/core/rawmidi.c:1163 __snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1163 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4608 Comm: syz-executor0 Not tainted 4.18.0-rc4+ #51 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:__snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1163 Code: ba 01 00 00 00 be 03 00 00 00 e8 88 4a 83 fb e8 43 fe a2 fb 89 d8 48 83 c4 18 5b 41 5c 41 5d 41 5e 41 5f 5d c3 e8 2d fe a2 fb <0f> 0b e9 72 fe ff ff e8 e1 dc e0 fb e9 1b ff ff ff bb ea ff ff ff RSP: 0018:ffff8801abb8f720 EFLAGS: 00010093 RAX: ffff8801d8b500c0 RBX: 0000000000000002 RCX: ffffffff85d90b82 RDX: 0000000000000000 RSI: ffffffff85d90d13 RDI: 0000000000000006 RBP: ffff8801abb8f760 R08: ffff8801d8b500c0 R09: ffffed003b128cb4 R10: ffffed003b128cb4 R11: ffff8801d89465a3 R12: ffff8801cd646240 R13: 0000000000000802 R14: 0000000000000002 R15: ffff8801ab48e5c0 snd_virmidi_output_trigger+0x3ae/0x6b0 sound/core/seq/seq_virmidi.c:194 snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [inline] snd_rawmidi_kernel_write1+0x519/0x700 sound/core/rawmidi.c:1288 snd_rawmidi_write+0x2ea/0xde0 sound/core/rawmidi.c:1338 __vfs_write+0x117/0x9f0 fs/read_write.c:485 vfs_write+0x1f8/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __ia32_sys_write+0x71/0xb0 fs/read_write.c:607 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f5fcb9 Code: 55 08 8b 88 64 cd ff ff 8b 98 68 cd ff ff 89 c8 85 d2 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 1c 24 c3 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f7f190ac EFLAGS: 00000282 ORIG_RAX: 0000000000000004 RAX: ffffffffffffffda RBX: 0000000000000006 RCX: 00000000200000c0 RDX: 000000000000ff4c RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix routing
by Russell King 14 Jul '18

14 Jul '18

Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.7.4

1 0

[alsa-devel] [sound:topic/hda-acomp 20/27] htmldocs: include/drm/i915_component.h:45: warning: Function parameter or member 'base' not described in 'i915_audio_component'
by kbuild test robot 13 Jul '18

13 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-acomp head: db43f0e9b348f78c88362d28c835d479730fbdb0 commit: 292c80bd71755c9f07fd14aee517121c72e13328 [20/27] drm/i915: Split audio component to a generic type reproduce: make htmldocs All warnings (new ones prefixed by >>): WARNING: convert(1) not found, for SVG to PDF conversion install ImageMagick (https://www.imagemagick.org) mm/mempool.c:228: warning: Function parameter or member 'pool' not described in 'mempool_init' Error: Cannot open file arch/x86/kernel/cpu/mtrr/main.c Error: Cannot open file arch/x86/kernel/cpu/mtrr/main.c WARNING: kernel-doc 'scripts/kernel-doc -rst -enable-lineno -export arch/x86/kernel/cpu/mtrr/main.c' failed with return code 2 include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ibss' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.connect' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.keys' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ie' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ie_len' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.bssid' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.ssid' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.default_key' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.default_mgmt_key' not described in 'wireless_dev' include/net/cfg80211.h:4279: warning: Function parameter or member 'wext.prev_bssid_valid' not described in 'wireless_dev' include/net/mac80211.h:2282: warning: Function parameter or member 'radiotap_timestamp.units_pos' not described in 'ieee80211_hw' include/net/mac80211.h:2282: warning: Function parameter or member 'radiotap_timestamp.accuracy' not described in 'ieee80211_hw' include/net/mac80211.h:955: warning: Function parameter or member 'control.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.rts_cts_rate_idx' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.use_rts' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.use_cts_prot' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.short_preamble' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.skip_table' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.jiffies' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.vif' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.hw_key' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.flags' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'control.enqueue_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'ack' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'ack.cookie' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ampdu_ack_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.ampdu_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.antenna' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.tx_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.is_valid_ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'status.status_driver_data' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'driver_rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'pad' not described in 'ieee80211_tx_info' include/net/mac80211.h:955: warning: Function parameter or member 'rate_driver_data' not described in 'ieee80211_tx_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg.signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'rx_stats_avg.chain_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.filtered' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.retry_failed' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.retry_count' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.lost_packets' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_tdls_pkt_time' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.msdu_retries' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.msdu_failed' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_ack' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.last_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.ack_signal_filled' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'status_stats.avg_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.packets' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.bytes' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.last_rate' not described in 'sta_info' net/mac80211/sta_info.h:588: warning: Function parameter or member 'tx_stats.msdu' not described in 'sta_info' kernel/sched/fair.c:3760: warning: Function parameter or member 'flags' not described in 'attach_entity_load_avg' include/linux/device.h:93: warning: bad line: this bus. include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.cb' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.poll' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_excl.active' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.cb' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.poll' not described in 'dma_buf' include/linux/dma-buf.h:307: warning: Function parameter or member 'cb_shared.active' not described in 'dma_buf' include/linux/dma-fence-array.h:54: warning: Function parameter or member 'work' not described in 'dma_fence_array' include/linux/gpio/driver.h:142: warning: Function parameter or member 'request_key' not described in 'gpio_irq_chip' include/linux/iio/hw-consumer.h:1: warning: no structured comments found include/linux/device.h:94: warning: bad line: this bus. include/linux/input/sparse-keymap.h:46: warning: Function parameter or member 'sw' not described in 'key_entry' include/linux/regulator/driver.h:227: warning: Function parameter or member 'resume_early' not described in 'regulator_ops' drivers/regulator/core.c:4465: warning: Excess function parameter 'state' description in 'regulator_suspend_late' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw0' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw1' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw2' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw3' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.eadm' not described in 'irb' drivers/usb/dwc3/gadget.c:510: warning: Excess function parameter 'dwc' description in 'dwc3_gadget_start_config' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_pin' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_unpin' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_res_obj' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_get_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_import_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_vmap' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_vunmap' not described in 'drm_driver' include/drm/drm_drv.h:610: warning: Function parameter or member 'gem_prime_mmap' not described in 'drm_driver' >> include/drm/i915_component.h:45: warning: Function parameter or member 'base' not described in 'i915_audio_component' drivers/gpu/drm/i915/i915_vma.h:48: warning: cannot understand function prototype: 'struct i915_vma ' drivers/gpu/drm/i915/i915_vma.h:1: warning: no structured comments found include/drm/tinydrm/tinydrm.h:34: warning: Function parameter or member 'fb_dirty' not described in 'tinydrm_device' drivers/gpu/drm/tinydrm/mipi-dbi.c:272: warning: Function parameter or member 'crtc_state' not described in 'mipi_dbi_enable_flush' drivers/gpu/drm/tinydrm/mipi-dbi.c:272: warning: Function parameter or member 'plane_state' not described in 'mipi_dbi_enable_flush' vim +45 include/drm/i915_component.h 58fddc28 Imre Deak 2015-01-08 @45 :::::: The code at line 45 was first introduced by commit :::::: 58fddc288b5cec192ad9eb9221da7ed14d974a27 drm/i915: add component support :::::: TO: Imre Deak <imre.deak(a)intel.com> :::::: CC: Daniel Vetter <daniel.vetter(a)ffwll.ch> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data()
by Dan Carpenter 13 Jul '18

13 Jul '18

The main thing is that the data->priv[] array has AFE_PORT_MAX elements so the > condition should be >=. But we may as well check for negative values as well just to be safe. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index e7f809e3ee1f..89df25e711c5 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1329,7 +1329,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; }

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[alsa-devel] [PATCH] ASoC: qdsp6: q6routing: off by one in routing_hw_params()
by Dan Carpenter 13 Jul '18

13 Jul '18

The data->port_data[] array has AFE_MAX_PORTS elements so the check should be >= instead of > or we write one element beyond the end of the array. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 35269b492761..0e7cf9e35d51 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -927,7 +927,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id];

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