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

[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 06 Nov '25

06 Nov '25

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

3 12

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

2 2

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

4 4

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

27 Oct '23

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

3 2

[alsa-devel] Attempting to understand odd snd-usb-audio code and behavior
by Keith A. Milner 21 Aug '20

21 Aug '20

Over the last few weeks I've been trying to dig through the snd-usb-audio code to try to get an understanding of it. I've used ftrace and also a local version which I have heavily polluted with kprintf(KERN_INFO statements so I can track what it's doing in practice. I've been playing with this with some USB audio interfaces I have some which work, and some which don't, and trying to understand the differences and, ultimately, what I can do to make the non-functioning ones work. I've got to the point where I have a fairly good understanding of some parts, but I'm struggling to understand why the code is behaving as it does in practice. My observation when plugging the USB device in is that, as expected, it tries to build the audio device, and add components to it including attaching capture and playback streams. It seems to go through this phase without too many surprises, and we get to the standard dmesg output: usbcore: registered new interface driver snd-usb-audio After this, it gets confusing. It seems to open and close the PCMs (e.g, snd_usb_capture_open) several times in a row. Each time it sets up the hardware info, checks and sets the format and then closes the PCM. I don't see anything obviously different between most of these cycles. Here's an example my logging output (hopefully this will make sense): Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Setting up hw info Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Checking format: Aug 22 20:04:16 kerneldev kernel: [ 923.140790] Interface number = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting index = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] format type = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] frame size = 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140791] Setting pt=125 ptmin=125 Aug 22 20:04:16 kerneldev kernel: [ 923.140792] Checking for unusual bit rates Aug 22 20:04:16 kerneldev kernel: [ 923.140794] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.140921] Called: pcm.c - snd_usb_hw_params for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Setting format: Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Bytes per period = 8816 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Frames per period = 1102 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Periods per buffer = 4 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Channels = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Rate = 44100 Aug 22 20:04:16 kerneldev kernel: [ 923.141279] Called: endpoint.c - snd_usb_add_endpoint of type 0 with direction 1 Aug 22 20:04:16 kerneldev kernel: [ 923.141292] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141293] Called: pcm.c - configure_endpoint Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Setting endpoint params for endpoint num 142 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] channels=2 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] period bytes=8816 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] frames per period=1102 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] periods per buffer=4 Aug 22 20:04:16 kerneldev kernel: [ 923.141296] Called: endpoint.c - data_ep_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141301] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142768] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142772] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142774] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142830] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142836] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142837] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142846] Called: pcm.c - snd_usb_capture_close for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146760] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146834] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146838] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146841] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146845] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146848] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Setting up hw info It seems to do this several times, before the behavior changing slightly, as it now starts to try to set the endpoint parameters and seems to leave the stream open. However, it then tries to open the stream and set the endpoint parameters again resulting in errors: Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Setting endpoint params for endpoint num 142 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] channels=2 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] period bytes=1768 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] frames per period=221 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] periods per buffer=593 Aug 22 20:22:00 kerneldev kernel: [ 1987.452947] usb 1-1: Unable to change format on ep #8e: already in use Maybe this is driven by hotplug or something, but it seems to be performing a lot of pointless activities. Eventually it seems to try to open the PCM playback or capture stream (with some devices it opens playback, with others capture) and then checks the pointer: Aug 22 20:22:00 kerneldev kernel: [ 1987.470194] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470656] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470657] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Pointer is 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473567] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473569] Pointer is 0 The above is from a non-working device. For a working device the pointer increments. Eventually this activity ends: Aug 22 20:22:05 kerneldev kernel: [ 1992.473575] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473576] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473578] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473602] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503287] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503288] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503323] Called: pcm.c - snd_usb_playback_close for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503656] Called: init.c - snd_card_file_remove This seems as if it might be some sort of test of the PCMs, to make sure they are working. Certainly in non-working devices the pcm_pointer stays at zero and in working devices it increments. It also seems to try harder with some devices than others, even though the device descriptors seem very similar. Can anyone explain any of this to me, particularly why it's exhibiting this behaviour. I've uploaded one of the logs from a non-working device to Pastebin if anyone wants to look at a full example: https://pastebin.com/uBZEuQ57 Also, is this the best way to approach this, or are there other tools/techniques that can help? I've experimented with ftrace and that was useful, but didn't give the level of detail I wanted. Cheers, Keith

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[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ALSA: usb-audio: fix PCM device order
by Alexander Tsoy 27 Jul '19

27 Jul '19

Some cards have alternate setting with non-PCM format as the first altsetting in the interface descriptors. This confuses userspace, since alsa-lib uses device 0 by default. So lets parse interfaces in two steps: 1. Parse altsettings with PCM formats. 2. Parse altsettings with non-PCM formats. This fixes at least following cards: - Audinst HUD-mx2 - Audinst HUD-mini Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/stream.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8e9548bc1f1a..b2ebeae91396 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -477,7 +477,8 @@ static struct uac2_output_terminal_descriptor * return NULL; } -int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no, + bool *has_non_pcm, bool non_pcm) { struct usb_device *dev; struct usb_interface *iface; @@ -634,6 +635,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) iface_no, altno); continue; } + if (fmt->bFormatType != UAC_FORMAT_TYPE_I) + *has_non_pcm = true; + if ((fmt->bFormatType == UAC_FORMAT_TYPE_I) == non_pcm) + continue; if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { dev_err(&dev->dev, @@ -740,3 +745,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return 0; } +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + int err; + bool has_non_pcm = false; + + /* parse PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false); + if (err < 0) + return err; + + if (has_non_pcm) { + /* parse non-PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true); + if (err < 0) + return err; + } + + return 0; +} + -- 2.13.0

3 2

[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/x] ASoC: replace platform to component
by Kuninori Morimoto 12 Feb '18

12 Feb '18

Hi Mark These are for replace "platform" to "component". It needs 3 steps I think. step1. "component" needs _add_ function, like "platform" step2. "component" level pcm_new/free support step3. replace platform to component We can get platform pointer from rtd->platform on current style. But these are very categorized pointer, in other words, non generic pointer. This cleanup is 1st step of merging all feature into component. I think new style wil be CPU + Codec + Platform = Component I think merging CPU into Component is already done. This patch set merges Platform feature into Component, and remove Platform. Merging Codec into Component is next step. This patch set will add new rtd connected component list. All CPU/Codec/Platform are connected to this list. And we can get each component pointer by using rtd and its driver name. Here, We can replace rtd->platform like below - struct device *dev = rtd->platform->dev; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRIVER_NAME); + struct device *dev = component->dev; - if(rtd->platform->pcm_new) - rtd->platform->pcm_new(yyy); + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; /* rtd connected CPU/Codec/Platform */ + if (component->pcm_new) + component->pcm_new(yyy); + } Now, only platform has pcm_new/free, but in new style, all component can have it. I will post step1/2/3 patch-set. step1 is normal patches. step2/3 are now still [RFC]. Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 4/6] alsa-lib:pcm: fix the hw_ptr update until the boundary available.
by sutar.mounesh＠gmail.com 19 Jan '18

19 Jan '18

From: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> For long time test case, the slave_hw_ptr will exceed the boundary and wraparound the slave_hw_ptr. This slave boundary wraparound will cause the rate->hw_ptr to wraparound irrespective of the rate->boundary availability and due to that the available size goes wrong. Hence, To get the correct available size, - Its necessary to increment the rate->hw_ptr upto the rate->boundary and then wraparound the rate->hw_ptr. - While handling fraction part of slave period, rounded value will be introduced by input_frames(). To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Signed-off-by: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 1f830dd..5bee77c 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -50,7 +50,7 @@ typedef struct _snd_pcm_rate snd_pcm_rate_t; struct _snd_pcm_rate { snd_pcm_generic_t gen; - snd_pcm_uframes_t appl_ptr, hw_ptr; + snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; @@ -563,14 +563,28 @@ static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t sl { snd_pcm_rate_t *rate = pcm->private_data; + snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; + snd_pcm_sframes_t last_slave_hw_ptr_frac; + if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; - /* FIXME: boundary overlap of slave hw_ptr isn't evaluated here! - * e.g. if slave rate is small... - */ - rate->hw_ptr = - (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size + - rate->ops.input_frames(rate->obj, slave_hw_ptr % rate->gen.slave->period_size); + + if (slave_hw_ptr_diff < 0) + slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ + else if (slave_hw_ptr_diff == 0) + return; + last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; + /* While handling fraction part fo slave period, rounded value will be introduced by input_frames(). + * To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and + * add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Hence, + * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - + * fractional part of last_slave_hw_ptr rounded value + + * fractional part of updated slave hw ptr's rounded value ] */ + rate->hw_ptr += ( + (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - + rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + + rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + rate->last_slave_hw_ptr = slave_hw_ptr; rate->hw_ptr %= pcm->boundary; } @@ -635,6 +649,7 @@ static int snd_pcm_rate_prepare(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; @@ -650,6 +665,7 @@ static int snd_pcm_rate_reset(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: nuc900: Fix a loop timeout test
by Dan Carpenter 11 Dec '17

11 Dec '17

We should be finishing the loop with timeout set to zero but because this is a post-op we exit with timeout == -1. Fixes: 1082e2703a2d ("ASoC: NUC900/audio: add nuc900 audio driver support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index b6615affe571..fde974d52bb2 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -67,7 +67,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, /* polling the AC_R_FINISH */ while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH) - && timeout--) + && --timeout) mdelay(1); if (!timeout) { @@ -121,7 +121,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* polling the AC_W_FINISH */ while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH) - && timeout--) + && --timeout) mdelay(1); if (!timeout)

2 1

[alsa-devel] Cherry Trail + RT5645 devices with a mono speaker ?
by Hans de Goede 08 Dec '17

08 Dec '17

Hi Pierre-Louis, Bard, I've been looking into getting some Cherry Trail + RT5645 devices with a mono speaker I have to work properly. Specifically the goal is to mix the right channel into the left output, so that sounds which are only played over the right channel do not get lost. I'm using this UCM file: https://github.com/plbossart/UCM/blob/master/byt-rt5640/HiFi Looking at the rtl5640 UCM file, getting the mono speaker to work (in a first simple attempt) should be as simple as replacing: cset "name='SPOR MIX SPKVOL R Switch' on" With cset "name='SPOL MIX SPKVOL R Switch' on" But this does not work, the speaker test sound for the right speaker is still silent (works with headphones). I've also looked into directly poking the RT5645_SPO_MIXER i2c register for testing, but when checking its value with the original unmodified UCM file: [root@localhost ~]# i2cget -y -f 1 0x1a 0x48 w 0x06c8 That is: 0xc806 as the output of i2cget "w" mode needs byteswapping. Note that both the RT5645_M_SV_L_SPM_L and RT5645_M_SV_R_SPM_L bits are already cleared, which is weird as this is before I've modified anything. Also the RT5645_M_SV_R_SPM_R bit is cleared, but that is expected. Even if I manually set reg 0x48 to 0x07c8 which AFAIK should enable output of both left and right channels on the left speaker I still only get sounds played on the left channel. Any insights / help with this would be very much welcome. Regards, Hans

2 7

[alsa-devel] [PATCH] ASoC: fsl_asrc: protect macro argument
by Stefan Agner 06 Dec '17

06 Dec '17

Protect macro argument with parentheses to avoid ambiguity. This fixes a warning seen with clang: warning: logical not is only applied to the left hand side of this comparison Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_asrc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..ec33dab4b909 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -57,7 +57,7 @@ #define REG_ASRDOC 0x74 #define REG_ASRDI(i) (REG_ASRDIA + (i << 3)) #define REG_ASRDO(i) (REG_ASRDOA + (i << 3)) -#define REG_ASRDx(x, i) (x == IN ? REG_ASRDI(i) : REG_ASRDO(i)) +#define REG_ASRDx(x, i) ((x) == IN ? REG_ASRDI(i) : REG_ASRDO(i)) #define REG_ASRIDRHA 0x80 #define REG_ASRIDRLA 0x84 -- 2.13.3

3 7

[alsa-devel] [PATCH] ASoC: fsl_ssi: Override bit clock rate based on slot number
by Nicolin Chen 26 Nov '17

26 Nov '17

The set_sysclk() now is used to override the output bit clock rate. But this is not a common way to implement a set_dai_sysclk(). And this creates a problem when a general machine driver (simple-card for example) tries to do set_dai_sysclk() by passing an input clock rate for the baud clock instead of setting the bit clock rate as fsl_ssi driver expected. So this patch solves this problem by firstly removing set_sysclk() since the hw_params() can calculate the bit clock rate. Secondly, in order not to break those TDM use cases which previously might have been using set_sysclk() to override the bit clock rate, this patch changes the driver to override it based on the slot number. The patch also removes an obsolete comment of the dir parameter. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 64598d1..3657c88 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -197,12 +197,12 @@ struct fsl_ssi_soc_data { * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for both FIFOs used * @fifo_deph: Depth of the SSI FIFOs + * @slots: number of slots * @rxtx_reg_val: Specific register settings for receive/transmit configuration * * @clk: SSI clock * @baudclk: SSI baud clock for master mode * @baudclk_streams: Active streams that are using baudclk - * @bitclk_freq: bitclock frequency set by .set_dai_sysclk * * @dma_params_tx: DMA transmit parameters * @dma_params_rx: DMA receive parameters @@ -233,12 +233,12 @@ struct fsl_ssi_private { bool use_dual_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; + unsigned int slots; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct clk *clk; struct clk *baudclk; unsigned int baudclk_streams; - unsigned int bitclk_freq; /* regcache for volatile regs */ u32 regcache_sfcsr; @@ -700,8 +700,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, * Note: This function can be only called when using SSI as DAI master * * Quick instruction for parameters: - * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels - * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + * freq: Output BCLK frequency = samplerate * 32 (fixed) * slots (or channels) */ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -716,9 +715,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, unsigned int freq; bool baudclk_is_used; - /* Prefer the explicitly set bitclock frequency */ - if (ssi_private->bitclk_freq) - freq = ssi_private->bitclk_freq; + /* Generate bit clock based on the slot or channel number */ + if (ssi_private->slots) + freq = ssi_private->slots * 32 * params_rate(hw_params); else freq = params_channels(hw_params) * 32 * params_rate(hw_params); @@ -805,16 +804,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, return 0; } -static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - - ssi_private->bitclk_freq = freq; - - return 0; -} - /** * fsl_ssi_hw_params - program the sample size * @@ -1121,6 +1110,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); + ssi_private->slots = slots; + return 0; } @@ -1191,7 +1182,6 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, - .set_sysclk = fsl_ssi_set_dai_sysclk, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- 2.7.4

4 10

[alsa-devel] LADSPA ALSA plugin
by Robert Bielik 26 Nov '17

26 Nov '17

Hi all, Started experimenting with using LADSPA plugins with the ALSA LADSPA PCM plugin, and it works nicely. However, it seems only the PCM is interface is exposed from the plugin. Question is if there is a way to expose the LADSPA plugin parameters directly to alsamixer ? Regards /Rob

1 1

Re: [alsa-devel] KASAN: use-after-free Read in snd_timer_user_info_compat
by Takashi Iwai 21 Nov '17

21 Nov '17

On Fri, 27 Oct 2017 10:34:01 +0200, syzbot wrote: > > Hello, > > syzkaller hit the following crash on > 33d930e59a98fa10a0db9f56c7fa2f21a4aef9b9 > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > BUG: KASAN: use-after-free in > snd_timer_user_info_compat.isra.6+0x46c/0x480 > sound/core/timer_compat.c:71 > Read of size 8 at addr ffff8801d0dd9d00 by task syz-executor0/3342 OK, this is a real bug, and obviously a forgotten code path from the previous fixes. The patch below should address it, but I checked only compile-tested, so far. Let me know if it really works. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: timer: Add missing mutex lock for compat ioctls The races among ioctl and other operations were protected by the commit af368027a49a ("ALSA: timer: Fix race among timer ioctls") and later fixes, but one code path was forgotten in the scenario: the 32bit compat ioctl. As syzkaller recently spotted, a very similar use-after-free may happen with the combination of compat ioctls. The fix is simply to apply the same ioctl_lock to the compat_ioctl callback, too. Fixes: af368027a49a ("ALSA: timer: Fix race among timer ioctls") Reference: http://lkml.kernel.org/r/089e082686ac9b482e055c832617@google.com Reported-by: syzbot <bot+e5f3c9783e7048a74233054febbe9f1bdf54b6da(a)syzkaller.appspotmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer_compat.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 6a437eb66115..59127b6ef39e 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -133,7 +133,8 @@ enum { #endif /* CONFIG_X86_X32 */ }; -static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) +static long __snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) { void __user *argp = compat_ptr(arg); @@ -153,7 +154,7 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE: case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: - return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + return __snd_timer_user_ioctl(file, cmd, (unsigned long)argp); case SNDRV_TIMER_IOCTL_GPARAMS32: return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: @@ -167,3 +168,15 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns } return -ENOIOCTLCMD; } + +static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, + unsigned long arg) +{ + struct snd_timer_user *tu = file->private_data; + long ret; + + mutex_lock(&tu->ioctl_lock); + ret = __snd_timer_user_ioctl_compat(file, cmd, arg); + mutex_unlock(&tu->ioctl_lock); + return ret; +} -- 2.14.2

2 1

[alsa-devel] [PATCH] SoC: intel: byt: Introduce new custom IN2 map
by Carlo Caione 21 Nov '17

21 Nov '17

From: Carlo Caione <carlo(a)endlessm.com> Introduce a new custom dapm routes map to quirk platforms with the internal mic connected to IN2P. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3076bfc0db5e..1dad5c98c9ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -37,6 +37,7 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, + BYT_RT5651_IN2_MAP, }; #define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(7, 0)) @@ -58,6 +59,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) + dev_info(dev, "quirk IN2_MAP enabled"); if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) dev_info(dev, "quirk DMIC enabled"); if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) @@ -143,7 +146,6 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"ssp2 Rx", NULL, "AIF1 Capture"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ - {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "LOUTL"}, @@ -151,15 +153,23 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { + {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, + {"IN2P", NULL, "Headset Mic"}, {"IN1P", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN2P", NULL, "Internal Mic"}, +}; + static const struct snd_kcontrol_new byt_rt5651_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -246,6 +256,10 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; + case BYT_RT5651_IN2_MAP: + custom_map = byt_rt5651_intmic_in2_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); + break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); -- 2.14.2

3 5

[alsa-devel] [PATCH] ALSA: line6: add support for POD HD DESKTOP
by Hans P. Möller Ebner 20 Nov '17

20 Nov '17

Add support for the Line6 POD HD500X multi effect processor for playback and capture (in/out audio) through USB. Signed-off-by: Hans P. Moller <hmoller(a)uc.cl> --- sound/usb/line6/podhd.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 956f847..ba4f71f 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -39,7 +39,8 @@ enum { LINE6_PODHD500_1, LINE6_PODX3, LINE6_PODX3LIVE, - LINE6_PODHD500X + LINE6_PODHD500X, + LINE6_PODHDDESKTOP }; struct usb_line6_podhd { @@ -377,6 +378,7 @@ static const struct usb_device_id podhd_id_table[] = { { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, + { LINE6_IF_NUM(0x4156, 0), .driver_info = LINE6_PODHDDESKTOP }, {} }; @@ -463,6 +465,18 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, + [LINE6_PODHDDESKTOP] = { + .id = "PODHDDESKTOP", + .name = "POD HDDESKTOP", + .capabilities = LINE6_CAP_CONTROL + | LINE6_CAP_PCM | LINE6_CAP_HWMON, + .altsetting = 1, + .ep_ctrl_r = 0x81, + .ep_ctrl_w = 0x01, + .ctrl_if = 1, + .ep_audio_r = 0x86, + .ep_audio_w = 0x02, + }, }; /* -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: via82xx: Use common error handling code in snd_via82xx_create()
by SF Markus Elfring 19 Nov '17

19 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 19:37:26 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/via82xx_modem.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 55f79b2599e7..8659776887b2 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1110,8 +1110,8 @@ static int snd_via82xx_create(struct snd_card *card, return err; if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) { - pci_disable_device(pci); - return -ENOMEM; + err = -ENOMEM; + goto disable_device; } spin_lock_init(&chip->reg_lock); @@ -1121,30 +1121,27 @@ static int snd_via82xx_create(struct snd_card *card, if ((err = pci_request_regions(pci, card->driver)) < 0) { kfree(chip); - pci_disable_device(pci); - return err; + goto disable_device; } chip->port = pci_resource_start(pci, 0); if (request_irq(pci->irq, snd_via82xx_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_via82xx_free(chip); - return -EBUSY; + err = -EBUSY; + goto free_chip; } chip->irq = pci->irq; if (ac97_clock >= 8000 && ac97_clock <= 48000) chip->ac97_clock = ac97_clock; synchronize_irq(chip->irq); - if ((err = snd_via82xx_chip_init(chip)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_via82xx_chip_init(chip); + if (err < 0) + goto free_chip; - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_via82xx_free(chip); - return err; - } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto free_chip; /* The 8233 ac97 controller does not implement the master bit * in the pci command register. IMHO this is a violation of the PCI spec. @@ -1153,6 +1150,14 @@ static int snd_via82xx_create(struct snd_card *card, *r_via = chip; return 0; + +disable_device: + pci_disable_device(pci); + goto exit; +free_chip: + snd_via82xx_free(chip); +exit: + return err; } -- 2.14.0

3 6

[alsa-devel] [PATCH 0/7] ALSA: Fine-tuning for several function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 24 Jan 2017 11:02:34 +0100 Some update suggestions were taken into account from static source code analysis. Markus Elfring (7): seq: Delete unnecessary checks in snd_seq_ioctl_query_subs() seq: Delete unnecessary checks in snd_seq_ioctl_get_subscription() seq: Delete unnecessary checks in snd_seq_ioctl_unsubscribe_port() seq: Delete unnecessary checks in snd_seq_ioctl_subscribe_port() seq: Adjust 16 function calls together with a variable assignment dmasound_core: Move two assignments for the variable "ret" in state_open() dmasound_core: Adjust six function calls together with a variable assignment sound/core/seq/seq_clientmgr.c | 163 ++++++++++++++++++++----------------- sound/oss/dmasound/dmasound_core.c | 33 +++++--- 2 files changed, 109 insertions(+), 87 deletions(-) -- 2.11.0

2 9

[alsa-devel] [PATCH 1/1] ASoC: bcm2835-i2s: Support 384k sample rate
by Clive Messer 10 Nov '17

10 Nov '17

Add support for sample rates up to 384k to bcm2835-i2s driver, with SNDRV_PCM_RATE_CONTINUOUS and specifying 8k min and 384k max, for both playback and capture. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..4a67db2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -578,7 +578,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -586,7 +588,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.5.5

3 2

[alsa-devel] [PATCH 00/14] soundwire: Add a new SoundWire subsystem
by Vinod Koul 10 Nov '17

10 Nov '17

So today we celebrate Diwali, the festival of lights in this part of the world. On this occasion I would like to add a new Linux subsystem. This patch series adds a new SoundWire subsystem which implements a new MIPI bus protocol 'SoundWire'. The SoundWire protocol is a robust, scalable, low complexity, low power, low latency, two-pin (clock and data) multi-drop bus that allows for the transfer of multiple audio streams and embedded control/commands. SoundWire provides synchronization capabilities and supports both PCM and PDM, multichannel data, isochronous and asynchronous modes. This series adds SoundWire Bus, IO transfers, DisCo (Discovery and Configuration) sysfs interface, regmap and Documentation summary This patch series is also available on git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git topic/patch_v1 This is first version of the patch which incorporates lots of changes based on the RFC series posted last year https://lkml.org/lkml/2016/10/21/395 Sanyog Kale (4): Documentation: Add SoundWire summary soundwire: Add SoundWire MIPI defined registers soundwire: Add Slave status handling helpers soundwire: cdns: Add sdw_master_ops and IO transfer support Vinod Koul (10): soundwire: Add SoundWire bus type soundwire: Add Master registration soundwire: Add MIPI DisCo property helpers soundwire: Add IO transfer regmap: Add SoundWire bus support soundwire: Add slave status handling soundwire: Add sysfs for SoundWire DisCo properties soundwire: cdns: Add cadence module soundwire: intel: Add Intel Master driver soundwire: intel: Add Intel init module Documentation/soundwire/summary.txt | 192 +++++++ drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw.c | 146 +++++ drivers/soundwire/Kconfig | 37 ++ drivers/soundwire/Makefile | 18 + drivers/soundwire/bus.c | 983 ++++++++++++++++++++++++++++++++ drivers/soundwire/bus.h | 119 ++++ drivers/soundwire/bus_type.c | 245 ++++++++ drivers/soundwire/cadence_master.c | 793 ++++++++++++++++++++++++++ drivers/soundwire/cadence_master.h | 94 +++ drivers/soundwire/intel.c | 395 +++++++++++++ drivers/soundwire/intel.h | 70 +++ drivers/soundwire/intel_init.c | 230 ++++++++ drivers/soundwire/mipi_disco.c | 421 ++++++++++++++ drivers/soundwire/slave.c | 173 ++++++ drivers/soundwire/sysfs.c | 396 +++++++++++++ include/linux/mod_devicetable.h | 7 + include/linux/regmap.h | 37 ++ include/linux/soundwire/sdw.h | 564 ++++++++++++++++++ include/linux/soundwire/sdw_intel.h | 72 +++ include/linux/soundwire/sdw_registers.h | 238 ++++++++ scripts/mod/devicetable-offsets.c | 5 + scripts/mod/file2alias.c | 16 + 26 files changed, 5259 insertions(+) create mode 100644 Documentation/soundwire/summary.txt create mode 100644 drivers/base/regmap/regmap-sdw.c create mode 100644 drivers/soundwire/Kconfig create mode 100644 drivers/soundwire/Makefile create mode 100644 drivers/soundwire/bus.c create mode 100644 drivers/soundwire/bus.h create mode 100644 drivers/soundwire/bus_type.c create mode 100644 drivers/soundwire/cadence_master.c create mode 100644 drivers/soundwire/cadence_master.h create mode 100644 drivers/soundwire/intel.c create mode 100644 drivers/soundwire/intel.h create mode 100644 drivers/soundwire/intel_init.c create mode 100644 drivers/soundwire/mipi_disco.c create mode 100644 drivers/soundwire/slave.c create mode 100644 drivers/soundwire/sysfs.c create mode 100644 include/linux/soundwire/sdw.h create mode 100644 include/linux/soundwire/sdw_intel.h create mode 100644 include/linux/soundwire/sdw_registers.h -- 2.7.4

9 93

[alsa-devel] [PATCH 0/9] ASoC: prepare for Component conversion - step 2
by Kuninori Morimoto 08 Nov '17

08 Nov '17

Hi Mark These are resend of one of "replace component" patch series. I had posted it as below. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To review from big picture, please check above patch-set. I will continue to resend remainings of it if this small patch-set were accepted Kuninori Morimoto (9): ASoC: snd_soc_component_driver has snd_pcm_ops ASoC: snd_soc_component_driver has snd_compr_ops ASoC: snd_soc_component_driver has pmdown_time ASoC: snd_soc_component_driver has endianness ASoC: snd_soc_component_driver has non_legacy_dai_naming ASoC: add snd_soc_component_read32 ASoC: add snd_soc_component_xxx_bias_level() ASoC: add snd_soc_component_cache_sync() ASoC: add snd_soc_dapm_kcontrol_component() include/sound/soc.h | 78 ++++++++ sound/soc/soc-compress.c | 461 +++++++++++++++++++++++++++++++++++++++++++---- sound/soc/soc-core.c | 83 +++++---- sound/soc/soc-io.c | 14 ++ sound/soc/soc-pcm.c | 378 ++++++++++++++++++++++++++++++++++++-- 5 files changed, 929 insertions(+), 85 deletions(-) -- 1.9.1

2 11

[alsa-devel] [PATCH v2 0/4] TDA1997x HDMI video receiver
by Tim Harvey 07 Nov '17

07 Nov '17

This is a v4l2 subdev driver supporting the TDA1997x HDMI video receiver. I've tested this on a Gateworks GW54xx with an IMX6Q which uses the TDA19971 with 16bits connected to the IMX6 CSI. For this configuration I've tested both 16bit YUV422 and 8bit BT656 mode. While the driver should support the TDA1993 I do not have one for testing. Further potential development efforts include: - CEC support - HDCP support - mbus format selection support for bus widths that support multiple formats - TDA19972 support (2 inputs) History: v2: - encorporate feedback into dt bindings - change audio dt bindings - implement dv timings enum/cap - remove deprecated g_mbus_config op - fix dv_query_timings - add EDID get/set handling - remove max-pixel-rate support - add audio codec DAI support - added media-ctl and v4l2-compliance details v1: - initial RFC Media device topology: # media-ctl -d /dev/media0 -p Media controller API version 4.13.0 Media device information ------------------------ driver imx-media model imx-media serial bus info hw revision 0x0 driver version 4.13.0 Device topology - entity 1: adv7180 2-0020 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 20004 device node name /dev/v4l-subdev0 pad0: Source [fmt:UYVY8_2X8/720x480 field:interlaced colorspace:smpte170m] -> "ipu2_csi1_mux":1 [] - entity 3: tda19971 2-0048 (1 pad, 1 link) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev1 pad0: Source [fmt:UYVY8_1X16/0x0] [dv.caps:BT.656/1120 min:640x480@13000000 max:1920x1080@165000000 stds:CEA-861,DMT caps:progressive] [dv.query:no-link] [dv.current:BT.656/1120 0x0p0 (0x0) stds: flags:] -> "ipu1_csi0_mux":1 [ENABLED] - entity 5: ipu1_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev2 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] - entity 9: ipu2_vdic (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev3 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] pad2: Source [fmt:AYUV8_1X32/640x480@1/60 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] - entity 13: ipu1_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev4 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_vdic":2 [] <- "ipu1_csi0":1 [] <- "ipu1_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf":0 [] - entity 17: ipu1_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev5 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpenc capture":0 [] - entity 20: ipu1_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video0 pad0: Sink <- "ipu1_ic_prpenc":1 [] - entity 26: ipu1_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev6 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu1_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prpvf capture":0 [] - entity 29: ipu1_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video1 pad0: Sink <- "ipu1_ic_prpvf":1 [] - entity 35: ipu2_ic_prp (3 pads, 5 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev7 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_vdic":2 [] <- "ipu2_csi0":1 [] <- "ipu2_csi1":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf":0 [] - entity 39: ipu2_ic_prpenc (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev8 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":1 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpenc capture":0 [] - entity 42: ipu2_ic_prpenc capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video2 pad0: Sink <- "ipu2_ic_prpenc":1 [] - entity 48: ipu2_ic_prpvf (2 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev9 pad0: Sink [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] <- "ipu2_ic_prp":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prpvf capture":0 [] - entity 51: ipu2_ic_prpvf capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video3 pad0: Sink <- "ipu2_ic_prpvf":1 [] - entity 57: ipu1_csi0 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev10 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu1_csi0_mux":2 [ENABLED] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi0 capture":0 [ENABLED] - entity 61: ipu1_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video4 pad0: Sink <- "ipu1_csi0":2 [ENABLED] - entity 67: ipu1_csi1 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev11 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_ic_prp":0 [] -> "ipu1_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu1_csi1 capture":0 [] - entity 71: ipu1_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video5 pad0: Sink <- "ipu1_csi1":2 [] - entity 77: ipu2_csi0 (3 pads, 3 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev12 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi0 capture":0 [] - entity 81: ipu2_csi0 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video6 pad0: Sink <- "ipu2_csi0":2 [] - entity 87: ipu2_csi1 (3 pads, 4 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev13 pad0: Sink [fmt:UYVY8_2X8/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range crop.bounds:(0,0)/640x480 crop:(0,0)/640x480 compose.bounds:(0,0)/640x480 compose:(0,0)/640x480] <- "ipu2_csi1_mux":2 [] pad1: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_ic_prp":0 [] -> "ipu2_vdic":0 [] pad2: Source [fmt:AYUV8_1X32/640x480@1/30 field:none colorspace:smpte170m xfer:709 ycbcr:601 quantization:lim-range] -> "ipu2_csi1 capture":0 [] - entity 91: ipu2_csi1 capture (1 pad, 1 link) type Node subtype V4L flags 0 device node name /dev/video7 pad0: Sink <- "ipu2_csi1":2 [] - entity 97: ipu1_csi0_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev14 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "tda19971 2-0048":0 [ENABLED] pad2: Source [fmt:unknown/0x0] -> "ipu1_csi0":0 [ENABLED] - entity 101: ipu2_csi1_mux (3 pads, 2 links) type V4L2 subdev subtype Unknown flags 0 device node name /dev/v4l-subdev15 pad0: Sink [fmt:unknown/0x0] pad1: Sink [fmt:unknown/0x0] <- "adv7180 2-0020":0 [] pad2: Source [fmt:unknown/0x0] -> "ipu2_csi1":0 [] v4l2-compliance test results: root@ventana:~# v4l2-compliance -d /dev/video6 Driver Info: Driver name : imx-media-captu Card type : imx-media-capture Bus info : platform:ipu2_csi0 Driver version: 4.13.0 Capabilities : 0x84200001 Video Capture Streaming Extended Pix Format Device Capabilities Device Caps : 0x04200001 Video Capture Streaming Extended Pix Format Compliance test for device /dev/video6 (not using libv4l2): Required ioctls: test VIDIOC_QUERYCAP: OK Allow for multiple opens: test second video open: OK test VIDIOC_QUERYCAP: OK test VIDIOC_G/S_PRIORITY: OK Debug ioctls: test VIDIOC_DBG_G/S_REGISTER: OK (Not Supported) test VIDIOC_LOG_STATUS: OK (Not Supported) Input ioctls: test VIDIOC_G/S_TUNER/ENUM_FREQ_BANDS: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_S_HW_FREQ_SEEK: OK (Not Supported) test VIDIOC_ENUMAUDIO: OK (Not Supported) fail: v4l2-test-input-output.cpp(418): G_INPUT not supported for a capture device test VIDIOC_G/S/ENUMINPUT: FAIL test VIDIOC_G/S_AUDIO: OK (Not Supported) Inputs: 0 Audio Inputs: 0 Tuners: 0 Output ioctls: test VIDIOC_G/S_MODULATOR: OK (Not Supported) test VIDIOC_G/S_FREQUENCY: OK (Not Supported) test VIDIOC_ENUMAUDOUT: OK (Not Supported) test VIDIOC_G/S/ENUMOUTPUT: OK (Not Supported) test VIDIOC_G/S_AUDOUT: OK (Not Supported) Outputs: 0 Audio Outputs: 0 Modulators: 0 Input/Output configuration ioctls: test VIDIOC_ENUM/G/S/QUERY_STD: OK (Not Supported) test VIDIOC_ENUM/G/S/QUERY_DV_TIMINGS: OK (Not Supported) test VIDIOC_DV_TIMINGS_CAP: OK (Not Supported) test VIDIOC_G/S_EDID: OK (Not Supported) Control ioctls: test VIDIOC_QUERY_EXT_CTRL/QUERYMENU: OK test VIDIOC_QUERYCTRL: OK test VIDIOC_G/S_CTRL: OK fail: v4l2-test-controls.cpp(574): g_ext_ctrls does not support count == 0 test VIDIOC_G/S/TRY_EXT_CTRLS: FAIL test VIDIOC_(UN)SUBSCRIBE_EVENT/DQEVENT: OK (Not Supported) test VIDIOC_G/S_JPEGCOMP: OK (Not Supported) Standard Controls: 0 Private Controls: 0 Format ioctls: test VIDIOC_ENUM_FMT/FRAMESIZES/FRAMEINTERVALS: OK test VIDIOC_G/S_PARM: OK test VIDIOC_G_FBUF: OK (Not Supported) test VIDIOC_G_FMT: OK test VIDIOC_TRY_FMT: OK test VIDIOC_S_FMT: OK test VIDIOC_G_SLICED_VBI_CAP: OK (Not Supported) test Cropping: OK (Not Supported) test Composing: OK (Not Supported) test Scaling: OK (Not Supported) Codec ioctls: test VIDIOC_(TRY_)ENCODER_CMD: OK (Not Supported) test VIDIOC_G_ENC_INDEX: OK (Not Supported) test VIDIOC_(TRY_)DECODER_CMD: OK (Not Supported) Buffer ioctls: test VIDIOC_REQBUFS/CREATE_BUFS/QUERYBUF: OK test VIDIOC_EXPBUF: OK Test input 0: Total: 42, Succeeded: 40, Failed: 2, Warnings: 0 Tim Harvey (4): MAINTAINERS: add entry for NXP TDA1997x driver media: dt-bindings: Add bindings for TDA1997X media: i2c: Add TDA1997x HDMI receiver driver ARM: dts: imx: Add TDA19971 HDMI Receiver to GW54xx .../devicetree/bindings/media/i2c/tda1997x.txt | 179 ++ MAINTAINERS | 8 + arch/arm/boot/dts/imx6q-gw54xx.dts | 102 + arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 29 +- drivers/media/i2c/Kconfig | 9 + drivers/media/i2c/Makefile | 1 + drivers/media/i2c/tda1997x.c | 3336 ++++++++++++++++++++ include/dt-bindings/media/tda1997x.h | 78 + include/media/i2c/tda1997x.h | 53 + 9 files changed, 3792 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/media/i2c/tda1997x.txt create mode 100644 drivers/media/i2c/tda1997x.c create mode 100644 include/dt-bindings/media/tda1997x.h create mode 100644 include/media/i2c/tda1997x.h -- 2.7.4

3 17

[alsa-devel] [INTERNAL][PATCH 0/7] ASoC: stm32: Add SAI synchronization support
by Olivier Moysan 02 Nov '17

02 Nov '17

This patch-set adds support of synchronization features for SAI interface. It also adds minor fixes and improvements. Olivier Moysan (7): ASoC: stm32: Add synchronization to SAI bindings ASoC: stm32: sai: Move static settings to DAI init ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Remove spurious IRQs on stop ASoC: stm32: sai: Fix get reset controller ASoC: stm32: sai: Add synchronization support .../devicetree/bindings/sound/st,stm32-sai.txt | 14 +- sound/soc/stm/stm32_sai.c | 162 ++++++++++++++++++++- sound/soc/stm/stm32_sai.h | 22 ++- sound/soc/stm/stm32_sai_sub.c | 156 +++++++++++++++++--- 4 files changed, 318 insertions(+), 36 deletions(-) -- 1.9.1

3 16

[alsa-devel] [PATCH] ASoC: fix build warning in soc-core.c
by Randy Dunlap 02 Nov '17

02 Nov '17

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc build error. A symbol that ends with an underscore character ('_') has special meaning in reST (reStructuredText), so add a '*' to prevent this error and to indicate that there are several of these values to choose from. ../sound/soc/soc-core.c:2799: ERROR: Unknown target name: "snd_soc_daifmt". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-414-rc7.orig/sound/soc/soc-core.c +++ lnx-414-rc7/sound/soc/soc-core.c @@ -2796,7 +2796,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_r /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @fmt: SND_SOC_DAIFMT_ format value. + * @fmt: SND_SOC_DAIFMT_* format value. * * Configures the DAI hardware format and clocking. */

2 1

[alsa-devel] [PATCH RESEND1 00/12] ALSA: vsnd: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 02 Nov '17

02 Nov '17

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Thank you, Oleksandr Resending because of rebase onto [2] + added missing patch [1] https://xenproject.org/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/log/?h=for-… Oleksandr Andrushchenko (12): ALSA: vsnd: Introduce Xen para-virtualized sound frontend driver ALSA: vsnd: Implement driver's probe/remove ALSA: vsnd: Implement Xen bus state handling ALSA: vsnd: Read sound driver configuration from Xen store ALSA: vsnd: Implement Xen event channel handling ALSA: vsnd: Implement handling of shared buffers ALSA: vsnd: Introduce ALSA virtual sound driver ALSA: vsnd: Initialize virtul sound card ALSA: vsnd: Add timer for period interrupt emulation ALSA: vsnd: Implement ALSA PCM operations ALSA: vsnd: Implement communication with backend ALSA: vsnd: Introduce Kconfig option to enable Xen PV sound sound/drivers/Kconfig | 12 + sound/drivers/Makefile | 2 + sound/drivers/xen-front.c | 2107 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2121 insertions(+) create mode 100644 sound/drivers/xen-front.c -- 2.7.4

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[alsa-devel] [Patch v6 0/7] Introduce framework for SLIMbus device drivers
by srinivas.kandagatla＠linaro.org 02 Nov '17

02 Nov '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> It's been very long time since there was any activity on the slimbus patches, Am currently working on getting Qualcomm DSP based audio working on top of mainline. Slimbus is one of the major component for getting any analog audio on QCOM SoC's. So am taking intiative to address the review comments on the older patchset and send it. I have tested this patch on IFC6410 board with wcd9310 codec. SLIMbus (Serial Low Power Interchip Media Bus) is a specification developed by MIPI (Mobile Industry Processor Interface) alliance. SLIMbus is a 2-wire implementation, which is used to communicate with peripheral components like audio-codec. SLIMbus uses Time-Division-Multiplexing to accommodate multiple data channels, and control channel. Control channel has messages to do device-enumeration, messages to send/receive control-data to/from slimbus devices, messages for port/channel management, and messages to do bandwidth allocation. Framework is introduced to support multiple instances of the bus (1 controller per bus), and multiple slave devices per controller. SPI and I2C frameworks, and comments from last time when I submitted the patches were referred-to while working on this framework. These patchsets introduce device-management, OF helpers, and messaging APIs, controller driver for Qualcomm's slimbus controller, and clock-pause feature for entering/exiting low-power mode for SLIMbus. Framework patches to do channel, port and bandwidth management are work-in-progress and will be sent out once these initial patches are accepted. These patchsets were tested on Qualcomm Snapdragon processor board using the controller driver, and a test slave device. Changes from V5 to V6: * aligned slim_driver_register more like other buses, suggested by Arnd. * removed boardinfo and add_device apis for now, suggested by Arnd * Few namespace cleanups suggested by Masami * merged of apis in to first patch as suggested by Arnd. * slimbus clients "compatible" name space made much inline with USB and PCIE, suggested by Rob and Arnd. * Removed memory allocations to controller drivers, as suggested by Arnd. * Various bindings comments addressed as suggested by Mark and others. * Added regmap interface so that codecs can write more generic code. * Added MAINTAINER file. Sagar Dharia (5): slimbus: Device management on SLIMbus slimbus: Add messaging APIs to slimbus framework slimbus: qcom: Add Qualcomm Slimbus controller driver slimbus: Add support for 'clock-pause' feature slimbus: qcom: Add runtime-pm support using clock-pause feature Srinivas Kandagatla (2): regmap: add SLIMBUS support MAINTAINERS: Add SLIMbus maintainer Documentation/devicetree/bindings/slimbus/bus.txt | 57 ++ .../devicetree/bindings/slimbus/slim-qcom-ctrl.txt | 43 ++ Documentation/slimbus/summary | 109 ++++ MAINTAINERS | 8 + drivers/Kconfig | 2 + drivers/Makefile | 1 + drivers/base/regmap/Kconfig | 4 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-slimbus.c | 89 +++ drivers/slimbus/Kconfig | 20 + drivers/slimbus/Makefile | 8 + drivers/slimbus/slim-core.c | 725 +++++++++++++++++++++ drivers/slimbus/slim-messaging.c | 509 +++++++++++++++ drivers/slimbus/slim-qcom-ctrl.c | 714 ++++++++++++++++++++ drivers/slimbus/slim-qcom.h | 64 ++ drivers/slimbus/slim-sched.c | 126 ++++ include/linux/mod_devicetable.h | 13 + include/linux/regmap.h | 18 + include/linux/slimbus.h | 512 +++++++++++++++ 19 files changed, 3023 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/bus.txt create mode 100644 Documentation/devicetree/bindings/slimbus/slim-qcom-ctrl.txt create mode 100644 Documentation/slimbus/summary create mode 100644 drivers/base/regmap/regmap-slimbus.c create mode 100644 drivers/slimbus/Kconfig create mode 100644 drivers/slimbus/Makefile create mode 100644 drivers/slimbus/slim-core.c create mode 100644 drivers/slimbus/slim-messaging.c create mode 100644 drivers/slimbus/slim-qcom-ctrl.c create mode 100644 drivers/slimbus/slim-qcom.h create mode 100644 drivers/slimbus/slim-sched.c create mode 100644 include/linux/slimbus.h -- 2.9.3

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[alsa-devel] [PATCH] dmaengine: imx-sdma: advertise all supported slave bus widths
by Lucas Stach 31 Oct '17

31 Oct '17

The commit [1] which changed imx-pcm-dma to derive the pcm_config from the attached dmaengine introduced a regression on the supported sample formats, as imx-sdma doesn't properly advertise all the supported slave bus widths. Fix this to restore full functionality of the imx sound subsystem. [1] 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config") Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- drivers/dma/imx-sdma.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index a67ec1bdc4e0..e374183f19d0 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -1851,8 +1851,12 @@ static int sdma_probe(struct platform_device *pdev) sdma->dma_device.device_prep_dma_cyclic = sdma_prep_dma_cyclic; sdma->dma_device.device_config = sdma_config; sdma->dma_device.device_terminate_all = sdma_disable_channel_with_delay; - sdma->dma_device.src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - sdma->dma_device.dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + sdma->dma_device.src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_1_BYTE); + sdma->dma_device.dst_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_1_BYTE); sdma->dma_device.directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); sdma->dma_device.residue_granularity = DMA_RESIDUE_GRANULARITY_SEGMENT; sdma->dma_device.device_issue_pending = sdma_issue_pending; -- 2.11.0

4 11

[alsa-devel] [PATCH] ASoC: rt5645: remove unexisting route on new rt5645
by Bard Liao 31 Oct '17

31 Oct '17

"SPOL MIX DAC R1 Switch" and "SPOL MIX SPKVOL R Switch" are only exist in the early version of rt5645. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e34f278..b7eef6a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -432,6 +432,7 @@ struct rt5645_priv { int jack_type; bool en_button_func; bool hp_on; + int v_id; }; static int rt5645_reset(struct snd_soc_codec *codec) @@ -2516,9 +2517,7 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPKVOL L", "Switch", "SPK MIXL" }, { "SPKVOL R", "Switch", "SPK MIXR" }, - { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, { "SPOL MIX", "DAC L1 Switch", "DAC L1" }, - { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, { "SPOL MIX", "SPKVOL L Switch", "SPKVOL L" }, { "SPOR MIX", "DAC R1 Switch", "DAC R1" }, { "SPOR MIX", "SPKVOL R Switch", "SPKVOL R" }, @@ -2707,6 +2706,11 @@ static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { { "DAC R2 Mux", "IF1 DAC", "RT5645 IF1 DAC2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5645_old_dapm_routes[] = { + { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -3363,6 +3367,11 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (rt5645->v_id < 3) { + snd_soc_dapm_add_routes(dapm, + rt5645_old_dapm_routes, + ARRAY_SIZE(rt5645_old_dapm_routes)); + } break; case CODEC_TYPE_RT5650: snd_soc_dapm_new_controls(dapm, @@ -3803,6 +3812,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_write(rt5645->regmap, RT5645_RESET, 0); + regmap_read(regmap, RT5645_VENDOR_ID, &val); + rt5645->v_id = val & 0xff; + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/5] ASoC: rsnd: small tidyups for Renesas sound
by Kuninori Morimoto 31 Oct '17

31 Oct '17

Hi Mark These are small tidyups for Renesas sound driver. Kuninori Morimoto (5): ASoC: rsnd: don't use io->mod[] directly ASoC: rsnd: tidyup rsnd_mod_next() for loop method ASoC: rsnd: NULL check is not needed for clk_unprepare() ASoC: rsnd: use snd_pcm_running() in rsnd_io_is_working() ASoC: rsnd: Don't check SSISR::DIRQ when Capture sound/soc/sh/rcar/core.c | 16 ++++++++-------- sound/soc/sh/rcar/ssi.c | 15 +++++++++------ 2 files changed, 17 insertions(+), 14 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/2] Enable TDM slot and fix SSP params for kabylake machine driver
by Harsha Priya 31 Oct '17

31 Oct '17

This patch series includes 2 patches for kabylake machine driver using max98927 and rt5663 codecs. The changes are: 1. Enable 4 tdm slots for IV feedback and 2 tdm slots for playback on max98927 2. Fix setting of SSP parameters to be based on FE and BE dai links. Naveen M (2): ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver sound/soc/intel/boards/kbl_rt5663_max98927.c | 55 ++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 3 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: Intel: kbl: fix jack name
by Vinod Koul 31 Oct '17

31 Oct '17

Commit d1c4cb447a7e ("ASoC: Intel: Skylake: Fix jack name format substitution") added Jack name but erroneously added a space as well, so remove the space in Jack name. Fixes: d1c4cb447a7e ("ASoC: Intel: Skylake: Fix jack name format substitution") Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 9cd0769ccd34..69ab55956492 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -605,7 +605,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { codec = pcm->codec_dai->codec; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], NULL, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: Retire OSS
by Takashi Iwai 31 Oct '17

31 Oct '17

Hi, below is a snippet of the patch I'm going to apply for finally dropping the OSS code. It's been already disabled for a few kernel versions, and we didn't get any complains. So, as discussed in the last audio mini-summit, we go forward now. Since the whole patch just removing a bunch of files is too big to post (and useless), I attach only a part of it: namely, the changelog, the diffstat, and the files that are modified (but not deleted). thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] sound: Retire OSS Since no complaints have been raised after disabling the build of OSS (Open Sound System) by the commit 31cbee6a5611 ("sound: Disable the build of OSS drivers"), let's finally drop the whole code and documentation. Some glue codes are still left intact since sound/oss/dmasound stuff remains -- which is an independent implementation solely for m68k, and it's not covered by ALSA yet. Also, a couple of API header files (linux/sound.h and linux/soundcard.h) are kept remaining as well, since the OSS API itself is still supported by ALSA OSS emulation, and applications can refer to these. Where we're at it, some help texts in the top-level Kconfig are adjusted, too (who still needs to specify I/O port in kbuild nowadays?). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/oss/ALS | 66 - Documentation/sound/oss/AudioExcelDSP16 | 101 - Documentation/sound/oss/CMI8330 | 152 -- Documentation/sound/oss/ESS | 34 - Documentation/sound/oss/ESS1868 | 55 - Documentation/sound/oss/Introduction | 459 ----- Documentation/sound/oss/MultiSound | 1137 ----------- Documentation/sound/oss/OPL3 | 6 - Documentation/sound/oss/Opti | 218 -- Documentation/sound/oss/PAS16 | 162 -- Documentation/sound/oss/PSS | 41 - Documentation/sound/oss/PSS-updates | 88 - Documentation/sound/oss/README.OSS | 1455 ------------- Documentation/sound/oss/README.modules | 106 - Documentation/sound/oss/README.ymfsb | 107 - Documentation/sound/oss/SoundPro | 105 - Documentation/sound/oss/Soundblaster | 53 - Documentation/sound/oss/Tropez+ | 26 - Documentation/sound/oss/VIBRA16 | 80 - Documentation/sound/oss/WaveArtist | 170 -- Documentation/sound/oss/btaudio | 92 - Documentation/sound/oss/mwave | 185 -- Documentation/sound/oss/oss-parameters.txt | 51 - Documentation/sound/oss/ultrasound | 30 - MAINTAINERS | 11 - sound/Kconfig | 33 +- sound/oss/CHANGELOG | 369 ---- sound/oss/Kconfig | 533 ----- sound/oss/Makefile | 107 - sound/oss/README.FIRST | 6 - sound/oss/ad1848.c | 3062 ---------------------------- sound/oss/ad1848.h | 24 - sound/oss/ad1848_mixer.h | 253 --- sound/oss/aedsp16.c | 1373 ------------- sound/oss/audio.c | 985 --------- sound/oss/bin2hex.c | 39 - sound/oss/coproc.h | 12 - sound/oss/dev_table.c | 256 --- sound/oss/dev_table.h | 390 ---- sound/oss/dmabuf.c | 1268 ------------ sound/oss/hex2hex.c | 101 - sound/oss/kahlua.c | 229 --- sound/oss/midi_ctrl.h | 22 - sound/oss/midi_synth.c | 712 ------- sound/oss/midi_synth.h | 47 - sound/oss/midibuf.c | 427 ---- sound/oss/mpu401.c | 1804 ---------------- sound/oss/mpu401.h | 11 - sound/oss/msnd.c | 413 ---- sound/oss/msnd.h | 278 --- sound/oss/msnd_classic.c | 3 - sound/oss/msnd_classic.h | 185 -- sound/oss/msnd_pinnacle.c | 1941 ------------------ sound/oss/msnd_pinnacle.h | 246 --- sound/oss/opl3.c | 1255 ------------ sound/oss/opl3_hw.h | 246 --- sound/oss/os.h | 45 - sound/oss/pas2.h | 20 - sound/oss/pas2_card.c | 458 ----- sound/oss/pas2_midi.c | 262 --- sound/oss/pas2_mixer.c | 327 --- sound/oss/pas2_pcm.c | 419 ---- sound/oss/pss.c | 1270 ------------ sound/oss/sb.h | 185 -- sound/oss/sb_audio.c | 1097 ---------- sound/oss/sb_card.c | 354 ---- sound/oss/sb_card.h | 149 -- sound/oss/sb_common.c | 1287 ------------ sound/oss/sb_ess.c | 1822 ----------------- sound/oss/sb_ess.h | 34 - sound/oss/sb_midi.c | 206 -- sound/oss/sb_mixer.c | 770 ------- sound/oss/sb_mixer.h | 105 - sound/oss/sequencer.c | 1661 --------------- sound/oss/sleep.h | 18 - sound/oss/sound_calls.h | 87 - sound/oss/sound_config.h | 144 -- sound/oss/sound_firmware.h | 29 - sound/oss/sound_timer.c | 327 --- sound/oss/soundcard.c | 733 ------- sound/oss/soundvers.h | 2 - sound/oss/swarm_cs4297a.c | 2781 ------------------------- sound/oss/sys_timer.c | 280 --- sound/oss/trix.c | 525 ----- sound/oss/tuning.h | 23 - sound/oss/uart401.c | 477 ----- sound/oss/uart6850.c | 361 ---- sound/oss/ulaw.h | 69 - sound/oss/v_midi.c | 290 --- sound/oss/v_midi.h | 14 - sound/oss/vidc.c | 557 ----- sound/oss/vidc.h | 63 - sound/oss/vidc_fill.S | 218 -- sound/oss/waveartist.c | 2043 ------------------- sound/oss/waveartist.h | 92 - 95 files changed, 1 insertion(+), 41223 deletions(-) delete mode 100644 Documentation/sound/oss/ALS delete mode 100644 Documentation/sound/oss/AudioExcelDSP16 delete mode 100644 Documentation/sound/oss/CMI8330 delete mode 100644 Documentation/sound/oss/ESS delete mode 100644 Documentation/sound/oss/ESS1868 delete mode 100644 Documentation/sound/oss/Introduction delete mode 100644 Documentation/sound/oss/MultiSound delete mode 100644 Documentation/sound/oss/OPL3 delete mode 100644 Documentation/sound/oss/Opti delete mode 100644 Documentation/sound/oss/PAS16 delete mode 100644 Documentation/sound/oss/PSS delete mode 100644 Documentation/sound/oss/PSS-updates delete mode 100644 Documentation/sound/oss/README.OSS delete mode 100644 Documentation/sound/oss/README.modules delete mode 100644 Documentation/sound/oss/README.ymfsb delete mode 100644 Documentation/sound/oss/SoundPro delete mode 100644 Documentation/sound/oss/Soundblaster delete mode 100644 Documentation/sound/oss/Tropez+ delete mode 100644 Documentation/sound/oss/VIBRA16 delete mode 100644 Documentation/sound/oss/WaveArtist delete mode 100644 Documentation/sound/oss/btaudio delete mode 100644 Documentation/sound/oss/mwave delete mode 100644 Documentation/sound/oss/oss-parameters.txt delete mode 100644 Documentation/sound/oss/ultrasound delete mode 100644 sound/oss/CHANGELOG delete mode 100644 sound/oss/Kconfig delete mode 100644 sound/oss/Makefile delete mode 100644 sound/oss/README.FIRST delete mode 100644 sound/oss/ad1848.c delete mode 100644 sound/oss/ad1848.h delete mode 100644 sound/oss/ad1848_mixer.h delete mode 100644 sound/oss/aedsp16.c delete mode 100644 sound/oss/audio.c delete mode 100644 sound/oss/bin2hex.c delete mode 100644 sound/oss/coproc.h delete mode 100644 sound/oss/dev_table.c delete mode 100644 sound/oss/dev_table.h delete mode 100644 sound/oss/dmabuf.c delete mode 100644 sound/oss/hex2hex.c delete mode 100644 sound/oss/kahlua.c delete mode 100644 sound/oss/midi_ctrl.h delete mode 100644 sound/oss/midi_synth.c delete mode 100644 sound/oss/midi_synth.h delete mode 100644 sound/oss/midibuf.c delete mode 100644 sound/oss/mpu401.c delete mode 100644 sound/oss/mpu401.h delete mode 100644 sound/oss/msnd.c delete mode 100644 sound/oss/msnd.h delete mode 100644 sound/oss/msnd_classic.c delete mode 100644 sound/oss/msnd_classic.h delete mode 100644 sound/oss/msnd_pinnacle.c delete mode 100644 sound/oss/msnd_pinnacle.h delete mode 100644 sound/oss/opl3.c delete mode 100644 sound/oss/opl3_hw.h delete mode 100644 sound/oss/os.h delete mode 100644 sound/oss/pas2.h delete mode 100644 sound/oss/pas2_card.c delete mode 100644 sound/oss/pas2_midi.c delete mode 100644 sound/oss/pas2_mixer.c delete mode 100644 sound/oss/pas2_pcm.c delete mode 100644 sound/oss/pss.c delete mode 100644 sound/oss/sb.h delete mode 100644 sound/oss/sb_audio.c delete mode 100644 sound/oss/sb_card.c delete mode 100644 sound/oss/sb_card.h delete mode 100644 sound/oss/sb_common.c delete mode 100644 sound/oss/sb_ess.c delete mode 100644 sound/oss/sb_ess.h delete mode 100644 sound/oss/sb_midi.c delete mode 100644 sound/oss/sb_mixer.c delete mode 100644 sound/oss/sb_mixer.h delete mode 100644 sound/oss/sequencer.c delete mode 100644 sound/oss/sleep.h delete mode 100644 sound/oss/sound_calls.h delete mode 100644 sound/oss/sound_config.h delete mode 100644 sound/oss/sound_firmware.h delete mode 100644 sound/oss/sound_timer.c delete mode 100644 sound/oss/soundcard.c delete mode 100644 sound/oss/soundvers.h delete mode 100644 sound/oss/swarm_cs4297a.c delete mode 100644 sound/oss/sys_timer.c delete mode 100644 sound/oss/trix.c delete mode 100644 sound/oss/tuning.h delete mode 100644 sound/oss/uart401.c delete mode 100644 sound/oss/uart6850.c delete mode 100644 sound/oss/ulaw.h delete mode 100644 sound/oss/v_midi.c delete mode 100644 sound/oss/v_midi.h delete mode 100644 sound/oss/vidc.c delete mode 100644 sound/oss/vidc.h delete mode 100644 sound/oss/vidc_fill.S delete mode 100644 sound/oss/waveartist.c delete mode 100644 sound/oss/waveartist.h diff --git a/MAINTAINERS b/MAINTAINERS index 2d3d750b19c0..01f0b76dcc06 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -527,11 +527,6 @@ W: http://ez.analog.com/community/linux-device-drivers S: Supported F: drivers/input/misc/adxl34x.c -AEDSP16 DRIVER -M: Riccardo Facchetti <fizban(a)tin.it> -S: Maintained -F: sound/oss/aedsp16.c - AF9013 MEDIA DRIVER M: Antti Palosaari <crope(a)iki.fi> L: linux-media(a)vger.kernel.org @@ -9199,12 +9194,6 @@ F: include/linux/dt-bindings/mux/ F: include/linux/mux/ F: drivers/mux/ -MULTISOUND SOUND DRIVER -M: Andrew Veliath <andrewtv(a)usa.net> -S: Maintained -F: Documentation/sound/oss/MultiSound -F: sound/oss/msnd* - MULTITECH MULTIPORT CARD (ISICOM) S: Orphan F: drivers/tty/isicom.c diff --git a/sound/Kconfig b/sound/Kconfig index d7d2aac9542e..b6d047985905 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -3,25 +3,7 @@ menuconfig SOUND depends on HAS_IOMEM help If you have a sound card in your computer, i.e. if it can say more - than an occasional beep, say Y. Be sure to have all the information - about your sound card and its configuration down (I/O port, - interrupt and DMA channel), because you will be asked for it. - - You want to read the Sound-HOWTO, available from - <http://www.tldp.org/docs.html#howto>. General information about - the modular sound system is contained in the files - <file:Documentation/sound/oss/Introduction>. The file - <file:Documentation/sound/oss/README.OSS> contains some slightly - outdated but still useful information as well. Newer sound - driver documentation is found in <file:Documentation/sound/alsa/*>. - - If you have a PnP sound card and you want to configure it at boot - time using the ISA PnP tools (read - <http://www.roestock.demon.co.uk/isapnptools/>), then you need to - compile the sound card support as a module and load that module - after the PnP configuration is finished. To do this, choose M here - and read <file:Documentation/sound/oss/README.modules>; the module - will be called soundcore. + than an occasional beep, say Y. if SOUND @@ -114,19 +96,6 @@ source "sound/synth/Kconfig" endif # SND -menuconfig SOUND_PRIME - tristate "Open Sound System (DEPRECATED)" - select SOUND_OSS_CORE - depends on BROKEN - help - Say 'Y' or 'M' to enable Open Sound System drivers. - -if SOUND_PRIME - -source "sound/oss/Kconfig" - -endif # SOUND_PRIME - endif # !UML endif # SOUND -- 2.14.2

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Re: [alsa-devel] possible deadlock in snd_seq_deliver_event
by Dmitry Vyukov 31 Oct '17

31 Oct '17

On Fri, Oct 27, 2017 at 10:09 AM, syzbot <bot+7feb8de6b4d6bf810cf098bef942cc387e79d0ad(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzkaller hit the following crash on > 2bd6bf03f4c1c59381d62c61d03f6cc3fe71f66e > git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-next.git/master > compiler: gcc (GCC) 7.1.1 20170620 > .config is attached > Raw console output is attached. > C reproducer is attached > syzkaller reproducer is attached. See https://goo.gl/kgGztJ > for information about syzkaller reproducers > > > ============================================ > WARNING: possible recursive locking detected > 4.14.0-rc1+ #88 Not tainted > -------------------------------------------- > syzkaller883997/2981 is trying to acquire lock: > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] deliver_to_subscribers > sound/core/seq/seq_clientmgr.c:666 [inline] > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > but task is already holding lock: > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] deliver_to_subscribers > sound/core/seq/seq_clientmgr.c:666 [inline] > (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > other info that might help us debug this: > Possible unsafe locking scenario: > > CPU0 > ---- > lock(&grp->list_mutex); > lock(&grp->list_mutex); > > *** DEADLOCK *** > > May be due to missing lock nesting notation > > 2 locks held by syzkaller883997/2981: > #0: (register_mutex#4){+.+.}, at: [<ffffffff83d60ada>] > odev_release+0x4a/0x70 sound/core/seq/oss/seq_oss.c:152 > #1: (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] > #1: (&grp->list_mutex){++++}, at: [<ffffffff83d4dd49>] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > > stack backtrace: > CPU: 1 PID: 2981 Comm: syzkaller883997 Not tainted 4.14.0-rc1+ #88 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:16 [inline] > dump_stack+0x194/0x257 lib/dump_stack.c:52 > print_deadlock_bug kernel/locking/lockdep.c:1797 [inline] > check_deadlock kernel/locking/lockdep.c:1844 [inline] > validate_chain kernel/locking/lockdep.c:2453 [inline] > __lock_acquire+0x1232/0x4620 kernel/locking/lockdep.c:3498 > lock_acquire+0x1d5/0x580 kernel/locking/lockdep.c:4002 > down_read+0x96/0x150 kernel/locking/rwsem.c:23 > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] > snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] > snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 > snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 > sound/core/seq/seq_clientmgr.c:621 > snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 > snd_seq_kernel_client_dispatch+0x11e/0x150 > sound/core/seq/seq_clientmgr.c:2309 > snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:150 [inline] > snd_seq_oss_midi_reset+0x44b/0x700 sound/core/seq/oss/seq_oss_midi.c:481 > snd_seq_oss_synth_reset+0x398/0x980 sound/core/seq/oss/seq_oss_synth.c:416 > snd_seq_oss_reset+0x6c/0x260 sound/core/seq/oss/seq_oss_init.c:448 > snd_seq_oss_release+0x71/0x120 sound/core/seq/oss/seq_oss_init.c:425 > odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 > __fput+0x333/0x7f0 fs/file_table.c:210 > ____fput+0x15/0x20 fs/file_table.c:244 > task_work_run+0x199/0x270 kernel/task_work.c:112 > exit_task_work include/linux/task_work.h:21 [inline] > do_exit+0xa52/0x1b40 kernel/exit.c:865 > do_group_exit+0x149/0x400 kernel/exit.c:968 > SYSC_exit_group kernel/exit.c:979 [inline] > SyS_exit_group+0x1d/0x20 kernel/exit.c:977 > entry_SYSCALL_64_fastpath+0x1f/0xbe > RIP: 0033:0x442c58 > RSP: 002b:00007ffd15d4f8d8 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 > RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000442c58 > RDX: 0000000000000000 RSI: 000000000000003c RDI: 0000000000000000 > RBP: 0000000000000082 R08: 00000000000000e7 R09: ffffffffffffffd0 > R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000401ca0 > R13: 0000000000401d30 R14 I've just re-reproduced this on upstream 15f859ae5c43c7f0a064ed92d33f7a5bc5de6de0 (Oct 26): ============================================ WARNING: possible recursive locking detected 4.14.0-rc6+ #10 Not tainted -------------------------------------------- a.out/3062 is trying to acquire lock: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 but task is already holding lock: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(&grp->list_mutex); lock(&grp->list_mutex); *** DEADLOCK *** May be due to missing lock nesting notation 2 locks held by a.out/3062: #0: (register_mutex#4){+.+.}, at: [<ffffffff83d3b5da>] odev_release+0x4a/0x70 sound/core/seq/oss/seq_oss.c:152 #1: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] #1: (&grp->list_mutex){++++}, at: [<ffffffff83d28879>] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 stack backtrace: CPU: 0 PID: 3062 Comm: a.out Not tainted 4.14.0-rc6+ #10 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:16 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:52 print_deadlock_bug kernel/locking/lockdep.c:1797 [inline] check_deadlock kernel/locking/lockdep.c:1844 [inline] validate_chain kernel/locking/lockdep.c:2445 [inline] __lock_acquire+0xed5/0x3d50 kernel/locking/lockdep.c:3490 lock_acquire+0x1d5/0x580 kernel/locking/lockdep.c:3994 down_read+0x96/0x150 kernel/locking/rwsem.c:23 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:666 [inline] snd_seq_deliver_event+0x279/0x790 sound/core/seq/seq_clientmgr.c:807 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] snd_seq_deliver_event+0x318/0x790 sound/core/seq/seq_clientmgr.c:807 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 dummy_input+0x2c4/0x400 sound/core/seq/seq_dummy.c:104 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 snd_seq_deliver_event+0x12c/0x790 sound/core/seq/seq_clientmgr.c:818 snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2313 snd_seq_oss_dispatch sound/core/seq/oss/seq_oss_device.h:150 [inline] snd_seq_oss_midi_reset+0x44b/0x700 sound/core/seq/oss/seq_oss_midi.c:481 snd_seq_oss_synth_reset+0x398/0x980 sound/core/seq/oss/seq_oss_synth.c:416 snd_seq_oss_reset+0x6c/0x260 sound/core/seq/oss/seq_oss_init.c:448 snd_seq_oss_release+0x71/0x120 sound/core/seq/oss/seq_oss_init.c:425 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:112 exit_task_work include/linux/task_work.h:21 [inline] do_exit+0x9b5/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 SYSC_exit_group kernel/exit.c:979 [inline] SyS_exit_group+0x1d/0x20 kernel/exit.c:977 entry_SYSCALL_64_fastpath+0x1f/0xbe RIP: 0033:0x437b99 RSP: 002b:00007ffe1b782328 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000437b99 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000082 R08: 000000000000003c R09: 00000000000000e7 R10: ffffffffffffffc0 R11: 0000000000000246 R12: 0000000000000000 R13: 0000000000401c00 R14: 0000000000401c90 R15: 0000000000000000 > --- > This bug is generated by a dumb bot. It may contain errors. > See https://goo.gl/tpsmEJ for details. > Direct all questions to syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. > Once a fix for this bug is committed, please reply to this email with: > #syz fix: exact-commit-title > To mark this as a duplicate of another syzbot report, please reply with: > #syz dup: exact-subject-of-another-report > If it's a one-off invalid bug report, please reply with: > #syz invalid > Note: if the crash happens again, it will cause creation of a new bug > report. > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/001a114089d614a0f4055c82d0…. > For more options, visit https://groups.google.com/d/optout.

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[alsa-devel] sound tree breaks make clean
by Daniel Vetter 31 Oct '17

31 Oct '17

Hi all, The following commit commit 60d4981ccf3603669c8de304619048ad5da707bd Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Oct 24 09:15:23 2017 +0200 sound: Retire OSS breaks $ make defconfig && make clean (just make clean in an already clean tree seems to work). We've taken the sound tree out of drm-tip and our CI for now to work around this. Thanks, Daniel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

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[alsa-devel] [PATCH v2] ASoC: wm8741: Add digital mute callback
by Sergej Sawazki 31 Oct '17

31 Oct '17

Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v2: * Refactored debug message. sound/soc/codecs/wm8741.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b8c1940..5571882 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -351,6 +351,17 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + dev_dbg(codec->dev, "wm8741_mute: mute = %d\n", mute); + + snd_soc_update_bits(codec, WM8741_VOLUME_CONTROL, + WM8741_SOFT_MASK, !!mute << WM8741_SOFT_SHIFT); + return 0; +} + #define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ @@ -364,6 +375,7 @@ static const struct snd_soc_dai_ops wm8741_dai_ops = { .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, + .digital_mute = wm8741_mute, }; static struct snd_soc_dai_driver wm8741_dai = { -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC fixes for v4.14
by Mark Brown 31 Oct '17

31 Oct '17

The following changes since commit bb176f67090ca54869fc1262c913aa69d2ede070: Linux 4.14-rc6 (2017-10-23 06:49:47 -0400) are available in the git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.14-rc6 for you to fetch changes up to 0bebd2f1bfcbf954a93b65d4b2848488fd049c78: Merge remote-tracking branches 'asoc/fix/topology' and 'asoc/fix/wm8998' into asoc-linus (2017-10-26 09:45:52 +0200) ---------------------------------------------------------------- ASoC: Fixes for v4.14 I've been quite lax in sending these due to conference season but here's a fairly large collection of ASoC updates. The one thing that's not device specific is Takashi's fix for races between delayed work and PCM destruction, otherwise everything is specific to an individual device. ---------------------------------------------------------------- Bard Liao (3): ASoC: rt5659: register power bit of LOUT Amp ASoC: rt5659: connect LOUT Amp with Charge Pump ASoC: rt5616: fix 0x91 default value Charles Keepax (1): ASoC: wm8998: Correct handling of input muxes Chintan Patel (1): ASoC: Intel: Skylake: Fix jack name format substitution Christophe JAILLET (2): ASoC: topology: Fix a potential NULL pointer dereference in 'soc_tplg_dapm_widget_denum_create()' ASoC: topology: Fix a potential memory leak in 'soc_tplg_dapm_widget_denum_create()' Christophe Jaillet (2): ASoC: samsung: i2s: Fix error handling path in i2s_set_sysclk() ASoC: davinci-mcasp: Fix an error handling path in 'davinci_mcasp_probe()' Colin Ian King (1): ASoC: Intel: Skylake: fix swapped order of function arguments dir and pin_index Hsin-Yu Chao (2): ASoC: rt5514-spi: check irq status to schedule data copy ASoC: rt5514: Revert Hotword Model control Jean-François Têtu (2): ASoC: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1 ASoC: codecs: msm8916-wcd-analog: fix micbias level John Keeping (1): ASoC: rockchip: i2s: fix unbalanced clk_disable Kuninori Morimoto (2): ASoC: rsnd: fix ADG flags ASoC: rcar: skip disabled-SSI nodes Marek Szyprowski (1): ASoC: samsung: i2s: disable secondary DAI until it gets fixed Mark Brown (8): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/msm8916' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/rockchip' into asoc-linus Merge remote-tracking branches 'asoc/fix/adau17x1', 'asoc/fix/davinci' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5514', 'asoc/fix/rt5616', 'asoc/fix/rt5659', 'asoc/fix/rt5663', 'asoc/fix/samsung' and 'asoc/fix/stm32' into asoc-linus Merge remote-tracking branches 'asoc/fix/topology' and 'asoc/fix/wm8998' into asoc-linus Olivier Moysan (3): ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: Fix get reset controller Pierre-Louis Bossart (2): ASoC: max98090: reduce verbosity on PLL unlock ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code Ricard Wanderlof (1): ASoC: adau17x1: Workaround for noise bug in ADC Takashi Iwai (1): ASoC: pcm: Sync delayed work before releasing resources oder_chiou(a)realtek.com (1): ASoC: rt5663: Change the dev getting function in rt5663_irq sound/soc/codecs/adau17x1.c | 24 ++++++- sound/soc/codecs/adau17x1.h | 2 + sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 7 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 +- sound/soc/codecs/rt5514-spi.c | 17 ++++- sound/soc/codecs/rt5514-spi.h | 3 + sound/soc/codecs/rt5514.c | 63 ---------------- sound/soc/codecs/rt5514.h | 3 - sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5659.c | 4 +- sound/soc/codecs/rt5663.c | 3 +- sound/soc/codecs/wm8998.c | 84 +++++----------------- sound/soc/davinci/davinci-mcasp.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 + sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/samsung/i2s.c | 6 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/ssi.c | 5 +- sound/soc/soc-pcm.c | 13 +++- sound/soc/soc-topology.c | 5 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 33 +++++---- 25 files changed, 134 insertions(+), 169 deletions(-)

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[alsa-devel] "PM / QoS: Fix device resume latency PM QoS" breaks sound
by Chris Clayton 31 Oct '17

31 Oct '17

Hi, I pulled the latestchanges from Linus' tree this evening and have found that with the new kernel, sound is not working on my laptop. More precisely, the built-in speakers don't produce any sound. Sound does work when I use ear-plugs in the headphone socket. It also works via a bluetooth speaker. I've bisected the problem and ended up at: 0cc2b4e5a020fc7f4d1795741c116c983e9467d7 is the first bad commit commit 0cc2b4e5a020fc7f4d1795741c116c983e9467d7 Author: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Date: Tue Oct 24 15:20:45 2017 +0200 PM / QoS: Fix device resume latency PM QoS The special value of 0 for device resume latency PM QoS means "no restriction", but there are two problems with that. First, device resume latency PM QoS requests with 0 as the value are always put in front of requests with positive values in the priority lists used internally by the PM QoS framework, causing 0 to be chosen as an effective constraint value. However, that 0 is then interpreted as "no restriction" effectively overriding the other requests with specific restrictions which is incorrect. Second, the users of device resume latency PM QoS have no way to specify that *any* resume latency at all should be avoided, which is an artificial limitation in general. To address these issues, modify device resume latency PM QoS to use S32_MAX as the "no constraint" value and 0 as the "no latency at all" one and rework its users (the cpuidle menu governor, the genpd QoS governor and the runtime PM framework) to follow these changes. Also add a special "n/a" value to the corresponding user space I/F to allow user space to indicate that it cannot accept any resume latencies at all for the given device. Fixes: 85dc0b8a4019 (PM / QoS: Make it possible to expose PM QoS latency constraints) Link: https://bugzilla.kernel.org/show_bug.cgi?id=197323 Reported-by: Reinette Chatre <reinette.chatre(a)intel.com> Tested-by: Reinette Chatre <reinette.chatre(a)intel.com> Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Acked-by: Alex Shi <alex.shi(a)linaro.org> Cc: All applicable <stable(a)vger.kernel.org> :040000 040000 f0c128ec799bb9894cfc5c341f88ad7bdfb15bac 9a2e8171ca47f864bd534cd9c160cce58449a889 M Documentation :040000 040000 0028ffec81675e686bdd621c0445d3e814d7980c 29db53c6356a6fed9c8bdbc2d6bc7bd56a96e529 M drivers :040000 040000 2e66b79bd2ffb4fcb00f04a69a0afe5c80d1d3f3 dd6d8e90b59389cd2bd8a0c92716d79d2eeb8268 M include With that change reverted, the speakers emit sound again. The audio devices identified by "lspci -vv" are as follows: 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: CLEVO/KAPOK Computer Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at f7f14000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit- Address: 00000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05) Subsystem: CLEVO/KAPOK Computer 8 Series/C220 Series Chipset High Definition Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f7f10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Let me know if I can provide any additional diagnostics, but please cc me because I'm not subscribed. Thanks Chris

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[alsa-devel] [asoc:topic/intel 41/42] sound/soc/intel/boards/bytcr_rt5651.c:325:2: error: implicit declaration of function 'rt5651_set_jack_detect'
by kbuild test robot 30 Oct '17

30 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 2fe30129b0a6a8ddf71c46c8fb4524cb89739bc0 commit: d9f8f9b2f3e2b95875cd59ae5c3276e4662e7513 [41/42] ASoC: intel: byt: Add headset jack config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout d9f8f9b2f3e2b95875cd59ae5c3276e4662e7513 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/bytcr_rt5651.c: In function 'byt_rt5651_init': >> sound/soc/intel/boards/bytcr_rt5651.c:325:2: error: implicit declaration of function 'rt5651_set_jack_detect' [-Werror=implicit-function-declaration] rt5651_set_jack_detect(codec, &priv->jack); ^~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/rt5651_set_jack_detect +325 sound/soc/intel/boards/bytcr_rt5651.c 255 256 static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) 257 { 258 struct snd_soc_card *card = runtime->card; 259 struct snd_soc_codec *codec = runtime->codec; 260 struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); 261 const struct snd_soc_dapm_route *custom_map; 262 int num_routes; 263 int ret; 264 265 card->dapm.idle_bias_off = true; 266 267 switch (BYT_RT5651_MAP(byt_rt5651_quirk)) { 268 case BYT_RT5651_IN1_MAP: 269 custom_map = byt_rt5651_intmic_in1_map; 270 num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); 271 break; 272 case BYT_RT5651_IN2_MAP: 273 custom_map = byt_rt5651_intmic_in2_map; 274 num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); 275 break; 276 default: 277 custom_map = byt_rt5651_intmic_dmic_map; 278 num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); 279 } 280 ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); 281 if (ret) 282 return ret; 283 284 ret = snd_soc_add_card_controls(card, byt_rt5651_controls, 285 ARRAY_SIZE(byt_rt5651_controls)); 286 if (ret) { 287 dev_err(card->dev, "unable to add card controls\n"); 288 return ret; 289 } 290 snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); 291 snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); 292 293 if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { 294 /* 295 * The firmware might enable the clock at 296 * boot (this information may or may not 297 * be reflected in the enable clock register). 298 * To change the rate we must disable the clock 299 * first to cover these cases. Due to common 300 * clock framework restrictions that do not allow 301 * to disable a clock that has not been enabled, 302 * we need to enable the clock first. 303 */ 304 ret = clk_prepare_enable(priv->mclk); 305 if (!ret) 306 clk_disable_unprepare(priv->mclk); 307 308 if (byt_rt5651_quirk & BYT_RT5651_MCLK_25MHZ) 309 ret = clk_set_rate(priv->mclk, 25000000); 310 else 311 ret = clk_set_rate(priv->mclk, 19200000); 312 313 if (ret) 314 dev_err(card->dev, "unable to set MCLK rate\n"); 315 } 316 317 ret = snd_soc_card_jack_new(runtime->card, "Headset", 318 SND_JACK_HEADSET, &priv->jack, 319 bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); 320 if (ret) { 321 dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); 322 return ret; 323 } 324 > 325 rt5651_set_jack_detect(codec, &priv->jack); 326 327 return ret; 328 } 329 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] timestamp update when no hw delta
by Henrik Eriksson 30 Oct '17

30 Oct '17

Hi I am looking into commit 3179f62001880e588e229db3006a59ad87b7792a ("ALSA: core: add .get_time_info"), which, if I understand the code correctly, changed the behaviour of timestamps. Before this change, snd_pcm_update_hw_ptr0() just returned if the no_delta_check showed that the hw_ptr had not changed. With this change it will update the timestamps before returning. Was this intended, and if so does anyone have any insights to why? The original behaviour to not update the timestamps when the hw position has not changed seems more correct to me, but I am new to this code. Regards, /henrik

2 2

[alsa-devel] [PATCH 0/3] ASoC: wm8741: Varios enhancements
by Sergej Sawazki 30 Oct '17

30 Oct '17

This patch series contains the following 3 patches for the WM8741 codec driver. 1) Adds digital mute callback. 2) Sets oversampling rate mode (OSR), required for sampling rates > 48kHz. 3) Sets MCLK/LRCLK sampling rate ratio (SR), avoids relying on auto-detection. Sergej Sawazki (3): ASoC: wm8741: Add digital mute callback ASoC: wm8741: Set OSR mode in hw_params() ASoC: wm8741: Set SR mode in hw_params() sound/soc/codecs/wm8741.c | 49 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 8 deletions(-) -- 2.7.4

3 15

[alsa-devel] [PATCH linux-next 1/1] ASoC: rsnd: ssiu: reset SSI_MODE register
by jiada_wang＠mentor.com 30 Oct '17

30 Oct '17

From: Jiada Wang <jiada_wang(a)mentor.com> register SSI_MODE is set when SSI works in TDM Extended or TDM (Ex-)Split mode, but it isn't reset after SSI stops. this will cause issue, if SSI starts to work in other modes which requie SSI_MODE to have different value. This patch resets SSI_MODE register in .stop callback, when last stream calls .stop. Fixes: 186fadc132f0 ("ASoC: rsnd: add TDM Extend Mode support") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d94875..eff9f3b 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -217,6 +217,8 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, if (rsnd_ssi_multi_slaves_runtime(io)) rsnd_mod_write(mod, SSI_CONTROL, 0); + rsnd_mod_write(mod, SSI_MODE, 0); + return 0; } -- 1.7.9.5

3 3

[alsa-devel] [PATCH] sound: fix kernel-doc build warning
by Randy Dunlap 30 Oct '17

30 Oct '17

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc build error. A symbol that ends with an underscore character ('_') has special meaning in reST (reStructuredText), so add a '*' to prevent this error and to indicate that there are several of these values to choose from. ../sound/core/jack.c:312: ERROR: Unknown target name: "snd_jack_btn". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/core/jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-414-rc7.orig/sound/core/jack.c +++ lnx-414-rc7/sound/core/jack.c @@ -310,7 +310,7 @@ EXPORT_SYMBOL(snd_jack_set_parent); * @type: Jack report type for this key * @keytype: Input layer key type to be reported * - * Map a SND_JACK_BTN_ button type to an input layer key, allowing + * Map a SND_JACK_BTN_* button type to an input layer key, allowing * reporting of keys on accessories via the jack abstraction. If no * mapping is provided but keys are enabled in the jack type then * BTN_n numeric buttons will be reported.

2 1

[alsa-devel] message size limit
by Jaroslav Kysela 29 Oct '17

29 Oct '17

Hello all, I increased the maximal message size for this alsa-devel list to 200kB (as we discussed with Takashi to allow bigger patches for review). Anyway, if you like to send something big, it's better to upload the data to a public place and provide only URL. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

[alsa-devel] [PATCH RESEND 0/2] Enable TDM slot and fix SSP params for kabylake machine driver
by Harsha Priya 27 Oct '17

27 Oct '17

Hi Mark, These are resend of an already acked patch series. I posted it a few weeks ago but there has been update on the base branch. I want to resend these. Kindly review and merge. This patch series includes 2 patches for kabylake machine driver using max98927 and rt5663 codecs. The changes are: 1. Enable 4 tdm slots for IV feedback and 2 tdm slots for playback on max98927 2. Fix setting of SSP parameters to be based on FE and BE dai links. Harsha Priya (2): ASoC: Intel: Enable tdm slots for max98927 ASoC: Intel: Fix setting of SSP parameters in Kabylake machine driver sound/soc/intel/boards/kbl_rt5663_max98927.c | 67 +++++++++++++++++++++++++--- 1 file changed, 60 insertions(+), 7 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH] conf/ucm: rt5651: Fix HiFi file name
by Carlo Caione 26 Oct '17

26 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> The conf HiFi file name is HiFi.conf, fix the name in the main configuration file. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf index 2dcc72cc..60798ce7 100644 --- a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf +++ b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf @@ -1,7 +1,7 @@ # Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 SectionUseCase."HiFi" { - File "HiFi" + File "HiFi.conf" Comment "Play HiFi quality Music" } -- 2.14.2

2 1

[alsa-devel] [asoc:for-next 16/17] sound/soc/codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq'
by kbuild test robot 26 Oct '17

26 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 2708a8339e928912307fe5cd1ae66d959caa7279 commit: b16351eba5e00bd7957fb16b98a152e5425ceb74 [16/17] Merge remote-tracking branches 'asoc/topic/sgtl5000', 'asoc/topic/stm32', 'asoc/topic/tas571x', 'asoc/topic/tfa9879' and 'asoc/topic/ts3a277e' into asoc-next config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout b16351eba5e00bd7957fb16b98a152e5425ceb74 # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): >> sound/soc/codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq' static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc/codecs/rt5651.c:1787:20: note: previous definition of 'rt5651_irq' was here static irqreturn_t rt5651_irq(int irq, void *data) ^ >> sound/soc/codecs/rt5651.c:1884:12: error: redefinition of 'rt5651_jack_detect' static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc/codecs/rt5651.c:1797:12: note: previous definition of 'rt5651_jack_detect' was here static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ >> sound/soc/codecs/rt5651.c:1920:13: error: redefinition of 'rt5651_jack_detect_work' static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc/codecs/rt5651.c:1833:13: note: previous definition of 'rt5651_jack_detect_work' was here static void rt5651_jack_detect_work(struct work_struct *work) ^ >> sound/soc/codecs/rt5651.c:1949:5: error: redefinition of 'rt5651_set_jack_detect' int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ sound/soc/codecs/rt5651.c:1862:5: note: previous definition of 'rt5651_set_jack_detect' was here int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ In file included from include/linux/linkage.h:6:0, from include/linux/kernel.h:6, from include/linux/list.h:8, from include/linux/module.h:9, from sound/soc/codecs/rt5651.c:12: >> include/linux/export.h:67:20: error: redefinition of '__kstrtab_rt5651_set_jack_detect' static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ >> sound/soc/codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:67:20: note: previous definition of '__kstrtab_rt5651_set_jack_detect' was here static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc/codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: error: redefinition of '__ksymtab_rt5651_set_jack_detect' static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ >> sound/soc/codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: note: previous definition of '__ksymtab_rt5651_set_jack_detect' was here static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc/codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ sound/soc/codecs/rt5651.c:1833:13: warning: 'rt5651_jack_detect_work' defined but not used [-Wunused-function] static void rt5651_jack_detect_work(struct work_struct *work) ^ -- sound/soc//codecs/rt5651.c:1874:20: error: redefinition of 'rt5651_irq' static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc//codecs/rt5651.c:1787:20: note: previous definition of 'rt5651_irq' was here static irqreturn_t rt5651_irq(int irq, void *data) ^ sound/soc//codecs/rt5651.c:1884:12: error: redefinition of 'rt5651_jack_detect' static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc//codecs/rt5651.c:1797:12: note: previous definition of 'rt5651_jack_detect' was here static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) ^ sound/soc//codecs/rt5651.c:1920:13: error: redefinition of 'rt5651_jack_detect_work' static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc//codecs/rt5651.c:1833:13: note: previous definition of 'rt5651_jack_detect_work' was here static void rt5651_jack_detect_work(struct work_struct *work) ^ sound/soc//codecs/rt5651.c:1949:5: error: redefinition of 'rt5651_set_jack_detect' int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ sound/soc//codecs/rt5651.c:1862:5: note: previous definition of 'rt5651_set_jack_detect' was here int rt5651_set_jack_detect(struct snd_soc_codec *codec, ^ In file included from include/linux/linkage.h:6:0, from include/linux/kernel.h:6, from include/linux/list.h:8, from include/linux/module.h:9, from sound/soc//codecs/rt5651.c:12: >> include/linux/export.h:67:20: error: redefinition of '__kstrtab_rt5651_set_jack_detect' static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:67:20: note: previous definition of '__kstrtab_rt5651_set_jack_detect' was here static const char __kstrtab_##sym[] \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: error: redefinition of '__ksymtab_rt5651_set_jack_detect' static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1959:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ include/linux/export.h:70:36: note: previous definition of '__ksymtab_rt5651_set_jack_detect' was here static const struct kernel_symbol __ksymtab_##sym \ ^ include/linux/export.h:100:25: note: in expansion of macro '___EXPORT_SYMBOL' #define __EXPORT_SYMBOL ___EXPORT_SYMBOL ^ include/linux/export.h:107:2: note: in expansion of macro '__EXPORT_SYMBOL' __EXPORT_SYMBOL(sym, "_gpl") ^ sound/soc//codecs/rt5651.c:1872:1: note: in expansion of macro 'EXPORT_SYMBOL_GPL' EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); ^ sound/soc//codecs/rt5651.c:1833:13: warning: 'rt5651_jack_detect_work' defined but not used [-Wunused-function] static void rt5651_jack_detect_work(struct work_struct *work) ^ vim +/rt5651_irq +1874 sound/soc/codecs/rt5651.c b4435130 Carlo Caione 2017-10-20 1786 80bbe4a3 Carlo Caione 2017-10-20 @1787 static irqreturn_t rt5651_irq(int irq, void *data) 80bbe4a3 Carlo Caione 2017-10-20 1788 { 80bbe4a3 Carlo Caione 2017-10-20 1789 struct rt5651_priv *rt5651 = data; 80bbe4a3 Carlo Caione 2017-10-20 1790 80bbe4a3 Carlo Caione 2017-10-20 1791 queue_delayed_work(system_power_efficient_wq, 80bbe4a3 Carlo Caione 2017-10-20 1792 &rt5651->jack_detect_work, msecs_to_jiffies(250)); 80bbe4a3 Carlo Caione 2017-10-20 1793 80bbe4a3 Carlo Caione 2017-10-20 1794 return IRQ_HANDLED; 80bbe4a3 Carlo Caione 2017-10-20 1795 } 80bbe4a3 Carlo Caione 2017-10-20 1796 80bbe4a3 Carlo Caione 2017-10-20 @1797 static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) 80bbe4a3 Carlo Caione 2017-10-20 1798 { 80bbe4a3 Carlo Caione 2017-10-20 1799 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); 80bbe4a3 Carlo Caione 2017-10-20 1800 int jack_type; 80bbe4a3 Carlo Caione 2017-10-20 1801 80bbe4a3 Carlo Caione 2017-10-20 1802 if (jack_insert) { 80bbe4a3 Carlo Caione 2017-10-20 1803 snd_soc_dapm_force_enable_pin(dapm, "LDO"); 80bbe4a3 Carlo Caione 2017-10-20 1804 snd_soc_dapm_sync(dapm); 80bbe4a3 Carlo Caione 2017-10-20 1805 80bbe4a3 Carlo Caione 2017-10-20 1806 snd_soc_update_bits(codec, RT5651_MICBIAS, 80bbe4a3 Carlo Caione 2017-10-20 1807 RT5651_MIC1_OVCD_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1808 RT5651_MIC1_OVTH_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1809 RT5651_PWR_CLK12M_MASK | 80bbe4a3 Carlo Caione 2017-10-20 1810 RT5651_PWR_MB_MASK, 80bbe4a3 Carlo Caione 2017-10-20 1811 RT5651_MIC1_OVCD_EN | 80bbe4a3 Carlo Caione 2017-10-20 1812 RT5651_MIC1_OVTH_600UA | b16351eb Mark Brown 2017-10-26 1813 RT5651_PWR_MB_PU | b16351eb Mark Brown 2017-10-26 1814 RT5651_PWR_CLK12M_PU); b16351eb Mark Brown 2017-10-26 1815 msleep(100); b16351eb Mark Brown 2017-10-26 1816 if (snd_soc_read(codec, RT5651_IRQ_CTRL2) & RT5651_MB1_OC_CLR) b16351eb Mark Brown 2017-10-26 1817 jack_type = SND_JACK_HEADPHONE; b16351eb Mark Brown 2017-10-26 1818 else b16351eb Mark Brown 2017-10-26 1819 jack_type = SND_JACK_HEADSET; b16351eb Mark Brown 2017-10-26 1820 snd_soc_update_bits(codec, RT5651_IRQ_CTRL2, b16351eb Mark Brown 2017-10-26 1821 RT5651_MB1_OC_CLR, 0); b16351eb Mark Brown 2017-10-26 1822 } else { /* jack out */ b16351eb Mark Brown 2017-10-26 1823 jack_type = 0; b16351eb Mark Brown 2017-10-26 1824 b16351eb Mark Brown 2017-10-26 1825 snd_soc_update_bits(codec, RT5651_MICBIAS, b16351eb Mark Brown 2017-10-26 1826 RT5651_MIC1_OVCD_MASK, b16351eb Mark Brown 2017-10-26 1827 RT5651_MIC1_OVCD_DIS); b16351eb Mark Brown 2017-10-26 1828 } b16351eb Mark Brown 2017-10-26 1829 b16351eb Mark Brown 2017-10-26 1830 return jack_type; b16351eb Mark Brown 2017-10-26 1831 } b16351eb Mark Brown 2017-10-26 1832 b16351eb Mark Brown 2017-10-26 1833 static void rt5651_jack_detect_work(struct work_struct *work) b16351eb Mark Brown 2017-10-26 1834 { b16351eb Mark Brown 2017-10-26 1835 struct rt5651_priv *rt5651 = b16351eb Mark Brown 2017-10-26 1836 container_of(work, struct rt5651_priv, jack_detect_work.work); b16351eb Mark Brown 2017-10-26 1837 b16351eb Mark Brown 2017-10-26 1838 int report, val = 0; b16351eb Mark Brown 2017-10-26 1839 b16351eb Mark Brown 2017-10-26 1840 if (!rt5651->codec) b16351eb Mark Brown 2017-10-26 1841 return; b16351eb Mark Brown 2017-10-26 1842 b16351eb Mark Brown 2017-10-26 1843 switch (rt5651->pdata.jd_src) { b16351eb Mark Brown 2017-10-26 1844 case RT5651_JD1_1: b16351eb Mark Brown 2017-10-26 1845 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x1000; b16351eb Mark Brown 2017-10-26 1846 break; b16351eb Mark Brown 2017-10-26 1847 case RT5651_JD1_2: b16351eb Mark Brown 2017-10-26 1848 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x2000; b16351eb Mark Brown 2017-10-26 1849 break; b16351eb Mark Brown 2017-10-26 1850 case RT5651_JD2: b16351eb Mark Brown 2017-10-26 1851 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x4000; b16351eb Mark Brown 2017-10-26 1852 break; b16351eb Mark Brown 2017-10-26 1853 default: b16351eb Mark Brown 2017-10-26 1854 break; b16351eb Mark Brown 2017-10-26 1855 } b16351eb Mark Brown 2017-10-26 1856 b16351eb Mark Brown 2017-10-26 1857 report = rt5651_jack_detect(rt5651->codec, !val); b16351eb Mark Brown 2017-10-26 1858 b16351eb Mark Brown 2017-10-26 1859 snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); b16351eb Mark Brown 2017-10-26 1860 } b16351eb Mark Brown 2017-10-26 1861 b16351eb Mark Brown 2017-10-26 1862 int rt5651_set_jack_detect(struct snd_soc_codec *codec, b16351eb Mark Brown 2017-10-26 1863 struct snd_soc_jack *hp_jack) b16351eb Mark Brown 2017-10-26 1864 { b16351eb Mark Brown 2017-10-26 1865 struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); b16351eb Mark Brown 2017-10-26 1866 b16351eb Mark Brown 2017-10-26 1867 rt5651->hp_jack = hp_jack; b16351eb Mark Brown 2017-10-26 1868 rt5651_irq(0, rt5651); 3ae08dc0 Bard Liao 2015-12-23 1869 3ae08dc0 Bard Liao 2015-12-23 1870 return 0; 3ae08dc0 Bard Liao 2015-12-23 1871 } b16351eb Mark Brown 2017-10-26 1872 EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); 3ae08dc0 Bard Liao 2015-12-23 1873 60d5a1a4 Carlo Caione 2017-10-20 @1874 static irqreturn_t rt5651_irq(int irq, void *data) 60d5a1a4 Carlo Caione 2017-10-20 1875 { 60d5a1a4 Carlo Caione 2017-10-20 1876 struct rt5651_priv *rt5651 = data; 60d5a1a4 Carlo Caione 2017-10-20 1877 60d5a1a4 Carlo Caione 2017-10-20 1878 queue_delayed_work(system_power_efficient_wq, 60d5a1a4 Carlo Caione 2017-10-20 1879 &rt5651->jack_detect_work, msecs_to_jiffies(250)); 60d5a1a4 Carlo Caione 2017-10-20 1880 60d5a1a4 Carlo Caione 2017-10-20 1881 return IRQ_HANDLED; 60d5a1a4 Carlo Caione 2017-10-20 1882 } 60d5a1a4 Carlo Caione 2017-10-20 1883 60d5a1a4 Carlo Caione 2017-10-20 @1884 static int rt5651_jack_detect(struct snd_soc_codec *codec, int jack_insert) 60d5a1a4 Carlo Caione 2017-10-20 1885 { 60d5a1a4 Carlo Caione 2017-10-20 1886 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); 60d5a1a4 Carlo Caione 2017-10-20 1887 int jack_type; 60d5a1a4 Carlo Caione 2017-10-20 1888 60d5a1a4 Carlo Caione 2017-10-20 1889 if (jack_insert) { 60d5a1a4 Carlo Caione 2017-10-20 1890 snd_soc_dapm_force_enable_pin(dapm, "LDO"); 60d5a1a4 Carlo Caione 2017-10-20 1891 snd_soc_dapm_sync(dapm); 60d5a1a4 Carlo Caione 2017-10-20 1892 60d5a1a4 Carlo Caione 2017-10-20 1893 snd_soc_update_bits(codec, RT5651_MICBIAS, 60d5a1a4 Carlo Caione 2017-10-20 1894 RT5651_MIC1_OVCD_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1895 RT5651_MIC1_OVTH_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1896 RT5651_PWR_CLK12M_MASK | 60d5a1a4 Carlo Caione 2017-10-20 1897 RT5651_PWR_MB_MASK, 60d5a1a4 Carlo Caione 2017-10-20 1898 RT5651_MIC1_OVCD_EN | 60d5a1a4 Carlo Caione 2017-10-20 1899 RT5651_MIC1_OVTH_600UA | 80bbe4a3 Carlo Caione 2017-10-20 1900 RT5651_PWR_MB_PU | 80bbe4a3 Carlo Caione 2017-10-20 1901 RT5651_PWR_CLK12M_PU); 80bbe4a3 Carlo Caione 2017-10-20 1902 msleep(100); 80bbe4a3 Carlo Caione 2017-10-20 1903 if (snd_soc_read(codec, RT5651_IRQ_CTRL2) & RT5651_MB1_OC_CLR) 80bbe4a3 Carlo Caione 2017-10-20 1904 jack_type = SND_JACK_HEADPHONE; 80bbe4a3 Carlo Caione 2017-10-20 1905 else 80bbe4a3 Carlo Caione 2017-10-20 1906 jack_type = SND_JACK_HEADSET; 80bbe4a3 Carlo Caione 2017-10-20 1907 snd_soc_update_bits(codec, RT5651_IRQ_CTRL2, 80bbe4a3 Carlo Caione 2017-10-20 1908 RT5651_MB1_OC_CLR, 0); 80bbe4a3 Carlo Caione 2017-10-20 1909 } else { /* jack out */ 80bbe4a3 Carlo Caione 2017-10-20 1910 jack_type = 0; 80bbe4a3 Carlo Caione 2017-10-20 1911 80bbe4a3 Carlo Caione 2017-10-20 1912 snd_soc_update_bits(codec, RT5651_MICBIAS, 80bbe4a3 Carlo Caione 2017-10-20 1913 RT5651_MIC1_OVCD_MASK, 80bbe4a3 Carlo Caione 2017-10-20 1914 RT5651_MIC1_OVCD_DIS); 80bbe4a3 Carlo Caione 2017-10-20 1915 } 80bbe4a3 Carlo Caione 2017-10-20 1916 80bbe4a3 Carlo Caione 2017-10-20 1917 return jack_type; 80bbe4a3 Carlo Caione 2017-10-20 1918 } 80bbe4a3 Carlo Caione 2017-10-20 1919 80bbe4a3 Carlo Caione 2017-10-20 @1920 static void rt5651_jack_detect_work(struct work_struct *work) 80bbe4a3 Carlo Caione 2017-10-20 1921 { 80bbe4a3 Carlo Caione 2017-10-20 1922 struct rt5651_priv *rt5651 = 80bbe4a3 Carlo Caione 2017-10-20 1923 container_of(work, struct rt5651_priv, jack_detect_work.work); 80bbe4a3 Carlo Caione 2017-10-20 1924 80bbe4a3 Carlo Caione 2017-10-20 1925 int report, val = 0; 80bbe4a3 Carlo Caione 2017-10-20 1926 80bbe4a3 Carlo Caione 2017-10-20 1927 if (!rt5651->codec) 80bbe4a3 Carlo Caione 2017-10-20 1928 return; 80bbe4a3 Carlo Caione 2017-10-20 1929 80bbe4a3 Carlo Caione 2017-10-20 1930 switch (rt5651->pdata.jd_src) { 80bbe4a3 Carlo Caione 2017-10-20 1931 case RT5651_JD1_1: 80bbe4a3 Carlo Caione 2017-10-20 1932 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x1000; 80bbe4a3 Carlo Caione 2017-10-20 1933 break; 80bbe4a3 Carlo Caione 2017-10-20 1934 case RT5651_JD1_2: 80bbe4a3 Carlo Caione 2017-10-20 1935 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x2000; 80bbe4a3 Carlo Caione 2017-10-20 1936 break; 80bbe4a3 Carlo Caione 2017-10-20 1937 case RT5651_JD2: 80bbe4a3 Carlo Caione 2017-10-20 1938 val = snd_soc_read(rt5651->codec, RT5651_INT_IRQ_ST) & 0x4000; 80bbe4a3 Carlo Caione 2017-10-20 1939 break; 80bbe4a3 Carlo Caione 2017-10-20 1940 default: 80bbe4a3 Carlo Caione 2017-10-20 1941 break; 80bbe4a3 Carlo Caione 2017-10-20 1942 } 80bbe4a3 Carlo Caione 2017-10-20 1943 80bbe4a3 Carlo Caione 2017-10-20 1944 report = rt5651_jack_detect(rt5651->codec, !val); 80bbe4a3 Carlo Caione 2017-10-20 1945 80bbe4a3 Carlo Caione 2017-10-20 1946 snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); 80bbe4a3 Carlo Caione 2017-10-20 1947 } 80bbe4a3 Carlo Caione 2017-10-20 1948 80bbe4a3 Carlo Caione 2017-10-20 @1949 int rt5651_set_jack_detect(struct snd_soc_codec *codec, 80bbe4a3 Carlo Caione 2017-10-20 1950 struct snd_soc_jack *hp_jack) 80bbe4a3 Carlo Caione 2017-10-20 1951 { 80bbe4a3 Carlo Caione 2017-10-20 1952 struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); 80bbe4a3 Carlo Caione 2017-10-20 1953 80bbe4a3 Carlo Caione 2017-10-20 1954 rt5651->hp_jack = hp_jack; 80bbe4a3 Carlo Caione 2017-10-20 1955 rt5651_irq(0, rt5651); 80bbe4a3 Carlo Caione 2017-10-20 1956 80bbe4a3 Carlo Caione 2017-10-20 1957 return 0; 80bbe4a3 Carlo Caione 2017-10-20 1958 } 80bbe4a3 Carlo Caione 2017-10-20 @1959 EXPORT_SYMBOL_GPL(rt5651_set_jack_detect); 80bbe4a3 Carlo Caione 2017-10-20 1960 :::::: The code at line 1874 was first introduced by commit :::::: 60d5a1a47b9a8381c08d2263b11ac9c757c87746 ASoC: rt5651: Enable jack detection on JD* pins :::::: TO: Carlo Caione <carlo(a)endlessm.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2 0/5] rt5651: Enable jack detection
by Carlo Caione 26 Oct '17

26 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Patchset to support jack detection for the rt5651 codec on the JD* pins and enable it on the KIANO cherry-trail laptop. Changelog: v2: - Added quirks for the KIANO laptop - Extended support to JD1_2 and JD2 pins - Reworked quirk logic in the codec driver - Codec driver can now differentiate between headphone and headset (thank you Bard Liao) Carlo Caione (5): ASoC: rt5651: Enable jack detection on JD* pins ASoC: intel: byt: Add headset jack ASoC: intel: byt: Enable IN2 map quirk for a KIANO laptop ASoC: rt5651: Rework quirk logic ASoC: rt5651: Enable JD1_1 quirk for KIANO laptopt include/sound/rt5651.h | 8 ++ sound/soc/codecs/rt5651.c | 215 +++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5651.h | 4 + sound/soc/intel/boards/bytcr_rt5651.c | 31 +++++ 4 files changed, 252 insertions(+), 6 deletions(-) -- 2.14.2

2 11

[alsa-devel] [PATCH] dt-bindings: sgtl5000: Remove leading zero from '@0a' notation
by Marco Franchi 26 Oct '17

26 Oct '17

Improve the binding example by removing the leading 0 from '@0a' notation, which fixes the following build warning: Warning (unit_address_format): Node /sgtl5000@0a unit name should not have leading 0s Signed-off-by: Marco Franchi <marco.franchi(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 7a73a9d..060cb4a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -37,7 +37,7 @@ VDDIO 1.8V 2.5V 3.3V Example: -codec: sgtl5000@0a { +codec: sgtl5000@a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; -- 2.7.4

2 2

Re: [alsa-devel] ALC700 in the Intel chipset
by Pshou 26 Oct '17

26 Oct '17

Hi alsa-devel ML: Could you merge this 000-alc700-intel.patch file to your ALSA driver? I changed "Intel debug" to "Intel Reference board". Best regards Pshou -----Original Message----- From: Takashi Iwai [mailto:tiwai@suse.de] Sent: Friday, September 01, 2017 4:13 PM To: Pshou Cc: CHLin; Kailang; Singh, Guneshwor O Subject: Re: ALC700 in the Intel chipset On Fri, 01 Sep 2017 10:06:13 +0200, Pshou wrote: > > Hi Takashi: > > This is 0000-alc700-intel.patch file. Thanks, it looks good now. One thing to be changed is the name "Intel debug". Maybe "Intel Reference board" or such would fit better. In anyway, could you submit the patch to alsa-devel ML? Once after it's submitted, I'll merge it. thanks, Takashi > > Best regards > pshou > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Tuesday, August 22, 2017 5:04 PM > To: CHLin > Cc: Pshou; Kailang; Singh, Guneshwor O > Subject: Re: ALC700 in the Intel chipset > > On Tue, 22 Aug 2017 10:05:18 +0200, > CHLin wrote: > > > > Hi Takashi > > > > ALC 700 needs this patch for jack detection function. > > Because ALC700's jack detect function defaults is disable. > > So alc700 needs pathc to enable jack detection function. > > OK, then could you update the patch description for giving more such information, prepare the proper patch in a form described in Documentation/SubmittingPatches with signed-off-by tags? > This is mandatory for merging to upstream. > > > thanks, > > Takashi > > > Thanks. > > > > ======================= > > Realtek Semiconductor Corp. > > 林家賢 CH_Lin > > System Design Dept II PCBG > > Tel: 886-3-578-0211 ext:13162 > > ======================= > > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Monday, August 21, 2017 6:23 PM > > To: Pshou > > Cc: Kailang; Singh, Guneshwor O; CHLin > > Subject: Re: ALC700 in the Intel chipset > > > > On Mon, 21 Aug 2017 10:05:25 +0200, > > Pshou wrote: > > > > > > Hi Tiwai: > > > > > > Nice to meet you. > > > > > > This is the case of intel. > > > Fixing Jack detection on ALC700 > > > > Could you give more information about this patch? > > For which platform and for what problem? At best, please add more information in the patch changelog itself. > > > > > > thanks, > > > > Takashi > > > > ------Please consider the environment before printing this e-mail.

2 5

[alsa-devel] [PATCH] sound: Convert timers to use timer_setup()
by Kees Cook 26 Oct '17

26 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. These are all the "mechanical" changes remaining in the sound subsystem. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/i2c/other/ak4117.c | 8 ++++---- sound/pci/ctxfi/cttimer.c | 7 +++---- sound/pci/echoaudio/midi.c | 10 +++++----- sound/pci/korg1212/korg1212.c | 7 +++---- sound/pci/rme9652/hdsp.c | 8 ++++---- sound/pci/rme9652/hdspm.c | 8 ++++---- sound/soc/codecs/rt5645.c | 7 +++---- sound/soc/omap/ams-delta.c | 4 ++-- 8 files changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 3ab099fb8c15..b923342cadf4 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -35,7 +35,7 @@ MODULE_LICENSE("GPL"); #define AK4117_ADDR 0x00 /* fixed address */ -static void snd_ak4117_timer(unsigned long data); +static void snd_ak4117_timer(struct timer_list *t); static void reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char val) { @@ -91,7 +91,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t chip->read = read; chip->write = write; chip->private_data = private_data; - setup_timer(&chip->timer, snd_ak4117_timer, (unsigned long)chip); + timer_setup(&chip->timer, snd_ak4117_timer, 0); for (reg = 0; reg < 5; reg++) chip->regmap[reg] = pgm[reg]; @@ -529,9 +529,9 @@ int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags) return res; } -static void snd_ak4117_timer(unsigned long data) +static void snd_ak4117_timer(struct timer_list *t) { - struct ak4117 *chip = (struct ak4117 *)data; + struct ak4117 *chip = from_timer(chip, t, timer); if (chip->init) return; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 8f945341720b..08e874e9a7f6 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -63,9 +63,9 @@ struct ct_timer { * system-timer-based updates */ -static void ct_systimer_callback(unsigned long data) +static void ct_systimer_callback(struct timer_list *t) { - struct ct_timer_instance *ti = (struct ct_timer_instance *)data; + struct ct_timer_instance *ti = from_timer(ti, t, timer); struct snd_pcm_substream *substream = ti->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct ct_atc_pcm *apcm = ti->apcm; @@ -93,8 +93,7 @@ static void ct_systimer_callback(unsigned long data) static void ct_systimer_init(struct ct_timer_instance *ti) { - setup_timer(&ti->timer, ct_systimer_callback, - (unsigned long)ti); + timer_setup(&ti->timer, ct_systimer_callback, 0); } static void ct_systimer_start(struct ct_timer_instance *ti) diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8c685ddb1a41..6045a115cffe 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -199,9 +199,9 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) -static void snd_echo_midi_output_write(unsigned long data) +static void snd_echo_midi_output_write(struct timer_list *t) { - struct echoaudio *chip = (struct echoaudio *)data; + struct echoaudio *chip = from_timer(chip, t, timer); unsigned long flags; int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; @@ -257,8 +257,8 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream spin_lock_irq(&chip->lock); if (up) { if (!chip->tinuse) { - setup_timer(&chip->timer, snd_echo_midi_output_write, - (unsigned long)chip); + timer_setup(&chip->timer, snd_echo_midi_output_write, + 0); chip->tinuse = 1; } } else { @@ -273,7 +273,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream spin_unlock_irq(&chip->lock); if (up && !chip->midi_full) - snd_echo_midi_output_write((unsigned long)chip); + snd_echo_midi_output_write(&chip->timer); } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 04cd71c74e5c..c7b007164c99 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -599,9 +599,9 @@ static void snd_korg1212_SendStopAndWait(struct snd_korg1212 *korg1212) } /* timer callback for checking the ack of stop request */ -static void snd_korg1212_timer_func(unsigned long data) +static void snd_korg1212_timer_func(struct timer_list *t) { - struct snd_korg1212 *korg1212 = (struct snd_korg1212 *) data; + struct snd_korg1212 *korg1212 = from_timer(korg1212, t, timer); unsigned long flags; spin_lock_irqsave(&korg1212->lock, flags); @@ -2189,8 +2189,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, init_waitqueue_head(&korg1212->wait); spin_lock_init(&korg1212->lock); mutex_init(&korg1212->open_mutex); - setup_timer(&korg1212->timer, snd_korg1212_timer_func, - (unsigned long)korg1212); + timer_setup(&korg1212->timer, snd_korg1212_timer_func, 0); korg1212->irq = -1; korg1212->clkSource = K1212_CLKIDX_Local; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 9f0f73875f01..1bff4b1b39cd 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1410,9 +1410,9 @@ static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream, spin_unlock_irqrestore (&hdsp->lock, flags); } -static void snd_hdsp_midi_output_timer(unsigned long data) +static void snd_hdsp_midi_output_timer(struct timer_list *t) { - struct hdsp_midi *hmidi = (struct hdsp_midi *) data; + struct hdsp_midi *hmidi = from_timer(hmidi, t, timer); unsigned long flags; snd_hdsp_midi_output_write(hmidi); @@ -1439,8 +1439,8 @@ static void snd_hdsp_midi_output_trigger(struct snd_rawmidi_substream *substream spin_lock_irqsave (&hmidi->lock, flags); if (up) { if (!hmidi->istimer) { - setup_timer(&hmidi->timer, snd_hdsp_midi_output_timer, - (unsigned long) hmidi); + timer_setup(&hmidi->timer, snd_hdsp_midi_output_timer, + 0); mod_timer(&hmidi->timer, 1 + jiffies); hmidi->istimer++; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f20d42714e4d..4c59983158e0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1946,9 +1946,9 @@ snd_hdspm_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) spin_unlock_irqrestore (&hdspm->lock, flags); } -static void snd_hdspm_midi_output_timer(unsigned long data) +static void snd_hdspm_midi_output_timer(struct timer_list *t) { - struct hdspm_midi *hmidi = (struct hdspm_midi *) data; + struct hdspm_midi *hmidi = from_timer(hmidi, t, timer); unsigned long flags; snd_hdspm_midi_output_write(hmidi); @@ -1976,8 +1976,8 @@ snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) spin_lock_irqsave (&hmidi->lock, flags); if (up) { if (!hmidi->istimer) { - setup_timer(&hmidi->timer, snd_hdspm_midi_output_timer, - (unsigned long) hmidi); + timer_setup(&hmidi->timer, + snd_hdspm_midi_output_timer, 0); mod_timer(&hmidi->timer, 1 + jiffies); hmidi->istimer++; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a98647ac497c..735f2d0033dd 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3340,9 +3340,9 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } -static void rt5645_btn_check_callback(unsigned long data) +static void rt5645_btn_check_callback(struct timer_list *t) { - struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + struct rt5645_priv *rt5645 = from_timer(rt5645, t, btn_check_timer); queue_delayed_work(system_power_efficient_wq, &rt5645->jack_detect_work, msecs_to_jiffies(5)); @@ -3934,8 +3934,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); + timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 6c49f3d6fd96..d40219678700 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -260,7 +260,7 @@ static bool cx81801_cmd_pending; static bool ams_delta_muted; static DEFINE_SPINLOCK(ams_delta_lock); -static void cx81801_timeout(unsigned long data) +static void cx81801_timeout(struct timer_list *unused) { int muted; @@ -349,7 +349,7 @@ static void cx81801_receive(struct tty_struct *tty, /* First modem response, complete setup procedure */ /* Initialize timer used for config pulse generation */ - setup_timer(&cx81801_timer, cx81801_timeout, 0); + timer_setup(&cx81801_timer, cx81801_timeout, 0); v253_ops.receive_buf(tty, cp, fp, count); -- 2.7.4 -- Kees Cook Pixel Security

3 3

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 25 Oct '17

25 Oct '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..12f383fd7c39 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/7] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Subhransu S. Prusty 25 Oct '17

25 Oct '17

For certain platforms clocks (mclk/sclk/fs) are required to be up before the stream start. Example: some codecs needs the mclk/sclk/fs to be enabled early for a successful clock synchronization. Some patforms require clock to be enabled at boot and be always ON By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. set_dma_control API can be used with different payload configuration. Modify the arguments to take configurable parameters. v1 -> v2 - Register parent clocks with skylake device. With the patch "clk: Add support for runtime PM" soon to be merged will help DSP to stay active on call to clock enable. Reference: (https://patchwork.kernel.org/patch/9911741/) - Fix the machine driver to enable clocks early for headphone playback path as well to fix a pop noise issue - Include the eve machine driver changes as well Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Jaikrishna Nemallapudi (5): ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments ASoC: Intel: Skylake: Parse nhlt to populate clock information ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock ASoC: Intel: Skylake: Register clock device and ops ASoC: Intel: Skylake: Add ssp clock driver Naveen M (1): ASoC: Intel: eve: Enable mclk and ssp sclk early sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 98 ++++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 97 +++++++ sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 56 ++++ sound/soc/intel/skylake/skl-messages.c | 109 +++++++- sound/soc/intel/skylake/skl-nhlt.c | 148 +++++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 288 +++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 124 +++++++++ sound/soc/intel/skylake/skl-topology.h | 4 +- sound/soc/intel/skylake/skl.c | 235 +++++++++++++++++ sound/soc/intel/skylake/skl.h | 12 + 12 files changed, 1173 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 1.9.1

3 10

[alsa-devel] [PATCH] sound: oss: waveartist: Convert timers to use timer_setup()
by Kees Cook 25 Oct '17

25 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: David Howells <dhowells(a)redhat.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/oss/waveartist.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 4f0c3a232e41..c051c8e6023e 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -151,6 +151,7 @@ static DEFINE_SPINLOCK(waveartist_lock); #define machine_is_netwinder() 0 #else static struct timer_list vnc_timer; +static int timer_nr_waveartist_devs; static void vnc_configure_mixer(struct wavnc_info *devc, unsigned int input_mask); static int vnc_private_ioctl(int dev, unsigned int cmd, int __user *arg); @@ -1414,8 +1415,8 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - setup_timer(&vnc_timer, vnc_slider_tick, - nr_waveartist_devs); + timer_nr_waveartist_devs = nr_waveartist_devs; + timer_setup(&vnc_timer, vnc_slider_tick, 0); mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); @@ -1799,11 +1800,11 @@ vnc_slider(struct wavnc_info *devc) } static void -vnc_slider_tick(unsigned long data) +vnc_slider_tick(struct timer_list *unused) { int next_timeout; - if (vnc_slider(adev_info + data)) + if (vnc_slider(adev_info + timer_nr_waveartist_devs)) next_timeout = 5; // mixer reported change else next_timeout = VNC_TIMER_PERIOD; -- 2.7.4 -- Kees Cook Pixel Security

2 5

[alsa-devel] [PATCH] conf/ucm: rt5651: Add UCM config for bytcr-rt5651
by Carlo Caione 25 Oct '17

25 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Add a UCM configuration for the rt5651 codec on Intel's Cherry-Trail platform. Adapted from [0]. [0] https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/bytcr-rt5651/HiFi.conf | 126 +++++++++++++++++++++++++ src/conf/ucm/bytcr-rt5651/Makefile.am | 4 + src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf | 138 ++++++++++++++++++++++++++++ 5 files changed, 270 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/bytcr-rt5651/HiFi.conf create mode 100644 src/conf/ucm/bytcr-rt5651/Makefile.am create mode 100644 src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf diff --git a/configure.ac b/configure.ac index fbcfa829..c16f2461 100644 --- a/configure.ac +++ b/configure.ac @@ -703,6 +703,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/bytcr-rt5651/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index ea9b3850..2bd4ebe7 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 skylake-rt286 VEYRON-I2S chtrt5645 DB410c broxton-rt298 bytcr-rt5651 diff --git a/src/conf/ucm/bytcr-rt5651/HiFi.conf b/src/conf/ucm/bytcr-rt5651/HiFi.conf new file mode 100644 index 00000000..b9da30f7 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/HiFi.conf @@ -0,0 +1,126 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 + +SectionVerb { + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo HiFi enable sequence" + ] + + DisableSequence [ + exec "echo HiFi disable sequence" + ] + + Value { + PlaybackPCM "hw:bytcrrt5651" + CapturePCM "hw:bytcrrt5651" + } +} + +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Headphones playback enable sequence" + cset "name='Headphone Switch' on" + cset "name='HPO L Playback Switch' on" + cset "name='HPO R Playback Switch' on" + cset "name='HP Playback Volume' 100" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Headphones playback disable sequence" + cset "name='Headphone Switch' off" + cset "name='HPO L Playback Switch' off" + cset "name='HPO R Playback Switch' off" + cset "name='HP Playback Volume' 0" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} + +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Speaker enable sequence" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Speaker disable sequence" + cset "name='Speaker Switch' off" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Handset Microphone" + + ConflictingDevice [ + "Mainmic" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Handset Microphone enable sequence" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Handset Microphone disable sequence" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + ] + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + } +} + + +SectionDevice."Mainmic" { + Comment "Main Microphone" + + ConflictingDevice [ + "Handset" + ] + + EnableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Main Microphone enable sequence" + cset "name='RECMIXL BST2 Switch' on" + cset "name='RECMIXR BST2 Switch' on" + ] + + DisableSequence [ + cdev "hw:bytcrrt5651" + exec "echo Main Microphone disable sequence" + cset "name='RECMIXL BST2 Switch' off" + cset "name='RECMIXR BST2 Switch' off" + ] + + Value { + CaptureChannels "2" + } +} diff --git a/src/conf/ucm/bytcr-rt5651/Makefile.am b/src/conf/ucm/bytcr-rt5651/Makefile.am new file mode 100644 index 00000000..95aec978 --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/bytcr-rt5651 +ucm_DATA = bytcr-rt5651.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf new file mode 100644 index 00000000..2dcc72cc --- /dev/null +++ b/src/conf/ucm/bytcr-rt5651/bytcr-rt5651.conf @@ -0,0 +1,138 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 + +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music" +} + +SectionDefaults [ + cdev "hw:bytcrrt5651" + + # media mixer settings + # compress + cset "name='media0_in Gain 0 Switch' on" + cset "name='media0_in Gain 0 Volume' 0" + # normal + cset "name='media1_in Gain 0 Switch' on" + cset "name='media1_in Gain 0 Volume' 0" + # swm loopback + cset "name='media2_in Gain 0 Switch' off" + cset "name='media2_in Gain 0 Volume' 0%" + # deep buffer + cset "name='media3_in Gain 0 Switch' on" + cset "name='media3_in Gain 0 Volume' 0" + cset "name='media0_out mix 0 media0_in Switch' on" + cset "name='media0_out mix 0 media1_in Switch' on" + cset "name='media0_out mix 0 media2_in Switch' off" + cset "name='media0_out mix 0 media3_in Switch' on" + cset "name='media1_out mix 0 media0_in Switch' off" + cset "name='media1_out mix 0 media1_in Switch' off" + cset "name='media1_out mix 0 media2_in Switch' off" + cset "name='media1_out mix 0 media3_in Switch' off" + cset "name='pcm0_in Gain 0 Switch' on" + cset "name='pcm0_in Gain 0 Volume' 0" + cset "name='pcm1_in Gain 0 Switch' off" + cset "name='pcm1_in Gain 0 Volume' 0%" + + # codec0_out settings + cset "name='codec_out0 mix 0 codec_in0 Switch' off" + cset "name='codec_out0 mix 0 codec_in1 Switch' off" + cset "name='codec_out0 mix 0 media_loop1_in Switch' off" + cset "name='codec_out0 mix 0 media_loop2_in Switch' off" + cset "name='codec_out0 mix 0 pcm0_in Switch' on" + cset "name='codec_out0 mix 0 pcm1_in Switch' off" + cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" + cset "name='codec_out0 Gain 0 Switch' on" + cset "name='codec_out0 Gain 0 Volume' 0" + + # input settings + # pcm1_out settings + cset "name='codec_in0 Gain 0 Switch' on" + cset "name='codec_in0 Gain 0 Volume' 0" + cset "name='pcm1_out mix 0 codec_in0 Switch' on" + cset "name='pcm1_out mix 0 codec_in1 Switch' off" + cset "name='pcm1_out mix 0 media_loop1_in Switch' off" + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='pcm1_out mix 0 pcm0_in Switch' off" + cset "name='pcm1_out mix 0 pcm1_in Switch' off" + cset "name='pcm1_out mix 0 sprot_loop_in Switch' off" + cset "name='pcm1_out Gain 0 Switch' on" + cset "name='pcm1_out Gain 0 Volume' 0" + + # disable codec_out1 + cset "name='codec_out1 mix 0 codec_in0 Switch' off" + cset "name='codec_out1 mix 0 codec_in1 Switch' off" + cset "name='codec_out1 mix 0 media_loop1_in Switch' off" + cset "name='codec_out1 mix 0 media_loop2_in Switch' off" + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='codec_out1 mix 0 pcm1_in Switch' off" + cset "name='codec_out1 mix 0 sprot_loop_in Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Volume' 0%" + + # disable codec_in1 + cset "name='codec_in1 Gain 0 Switch' off" + cset "name='codec_in1 Gain 0 Volume' 0%" + + # disable all loops + cset "name='media_loop1_out mix 0 codec_in0 Switch' off" + cset "name='media_loop1_out mix 0 codec_in1 Switch' off" + cset "name='media_loop1_out mix 0 media_loop1_in Switch' off" + cset "name='media_loop1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop1_out mix 0 pcm0_in Switch' off" + cset "name='media_loop1_out mix 0 pcm1_in Switch' off" + cset "name='media_loop1_out mix 0 sprot_loop_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + cset "name='media_loop2_out mix 0 codec_in1 Switch' off" + cset "name='media_loop2_out mix 0 media_loop1_in Switch' off" + cset "name='media_loop2_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 pcm0_in Switch' off" + cset "name='media_loop2_out mix 0 pcm1_in Switch' off" + cset "name='media_loop2_out mix 0 sprot_loop_in Switch' off" + cset "name='sprot_loop_out mix 0 codec_in0 Switch' off" + cset "name='sprot_loop_out mix 0 codec_in1 Switch' off" + cset "name='sprot_loop_out mix 0 media_loop1_in Switch' off" + cset "name='sprot_loop_out mix 0 media_loop2_in Switch' off" + cset "name='sprot_loop_out mix 0 pcm0_in Switch' off" + cset "name='sprot_loop_out mix 0 pcm1_in Switch' off" + cset "name='sprot_loop_out mix 0 sprot_loop_in Switch' off" + + # default state + cset "name='ADC Capture Volume' 126" + cset "name='HP Playback Volume' 29" + cset "name='HPO L Playback Switch' on" + cset "name='HPO MIX DAC1 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPO R Playback Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + cset "name='IF1 ASRC Switch' on" + cset "name='IN Capture Volume' 27" + cset "name='IN2 Boost' 1" + cset "name='IN1 Boost' 1" + cset "name='LOUT L Playback Switch' on" + cset "name='LOUT MIX DAC L1 Switch' on" + cset "name='LOUT MIX DAC R1 Switch' on" + cset "name='LOUT R Playback Switch' on" + cset "name='OUT Playback Volume' 33" + cset "name='Stereo DAC MIXL DAC L1 Switch' on" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC2 Switch' on" + cset "name='Stereo2 ADC MIXL ADC1 Switch' on" + cset "name='Stereo2 ADC MIXL ADC2 Switch' on" + cset "name='Stereo2 ADC MIXR ADC1 Switch' on" + cset "name='Stereo2 ADC MIXR ADC2 Switch' on" + + # input + cset "name='Stereo1 ADC L1 Mux' ADC" + cset "name='Stereo1 ADC R1 Mux' ADC" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='RECMIXL BST3 Switch' off" + cset "name='RECMIXR BST3 Switch' off" + cset "name='RECMIXL BST2 Switch' off" + cset "name='RECMIXR BST2 Switch' off" +] -- 2.14.2

3 5

[alsa-devel] [PATCH] ALSA: sb: Minor optimization / fix of timer usage in sb8_midi.c
by Takashi Iwai 25 Oct '17

25 Oct '17

Currently the SB8 MIDI code sets up the timer object at each time before scheduling it at trigger callback, but basically this is superfluous once after set up. Also, the code misses the del_timer_sync() call that may leave a race condition for use-after-free. This patch addresses these issues, moving timer_setup() to snd_sb8dsp_midi(), and adding the del_timer_sync() call at snd_sb8dsp_midi_output_trigger() to make sure. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/sb/sb8_midi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index 05ba2c6b1a8c..4affdcb78f72 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -138,6 +138,7 @@ static int snd_sb8dsp_midi_output_close(struct snd_rawmidi_substream *substream) struct snd_sb *chip; chip = substream->rmidi->private_data; + del_timer_sync(&chip->midi_timer); spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~(SB_OPEN_MIDI_OUTPUT | SB_OPEN_MIDI_OUTPUT_TRIGGER); chip->midi_substream_output = NULL; @@ -230,8 +231,6 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre spin_lock_irqsave(&chip->open_lock, flags); if (up) { if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) { - timer_setup(&chip->midi_timer, - snd_sb8dsp_midi_output_timer, 0); mod_timer(&chip->midi_timer, 1 + jiffies); chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER; } @@ -274,6 +273,7 @@ int snd_sb8dsp_midi(struct snd_sb *chip, int device) if (chip->hardware >= SB_HW_20) rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = chip; + timer_setup(&chip->midi_timer, snd_sb8dsp_midi_output_timer, 0); chip->rmidi = rmidi; return 0; } -- 2.14.2

1 0

[alsa-devel] [RFCv3][PATCH 00/39] axfer: rewrite aplay, adding 'timer-based scheduling' option
by Takashi Sakamoto 25 Oct '17

25 Oct '17

Hi, This 3rd RFC patchset updates my previous one: [alsa-devel] [RFCv2][PATCH 00/38] alsa-utils: axfer: rewrite aplay http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125574.… For aim of this rewrite and its intension, please refer to my former posts. In this version, I add an option to support 'timer-based scheduling' scenario, which PulseAudio developers introduced. Please refer to patch 32: * axfer: add support for timer-based MMAP operation You can use this mode by adding '-sched-type=timer' option into command line. The other difference from my previous one: - fix some bugs to handle non-interleaved buffer on mapped page frame. - use snd_pcm_status() instead of snd_pcm_state() to execute hwsync. - minor fixes. In my plan, this is the last version including below patches. In next post, options added by below patches are categorized as obsoleted: * axfer: add an option to support volume unit meter * axfer: add a unit test for vumeter calculation * axfer: add an option for formatted filename * axfer: add an option to handle key events * axfer: add a parser for channel map API * axfer: add a feature to generate a file for process id Additionally, for next post, I'll prepare for help messages and man for this program. Regards Takashi Sakamoto (39): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a parser for command-line options axfer: add a common interface to transfer data frames axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add a common interface of waiter for I/O event notification axfer: add support of waiter for poll(2) axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add an implementation of waiter for epoll(7) axfer: add an option for waiter type axfer: add support for timer-based MMAP operation axfer: add an option to support volume unit meter axfer: add a unit test for vumeter calculation axfer: add an option for formatted filename axfer: add an option to handle key events axfer: add a parser for channel map API axfer: add a feature to generate a file for process id axfer: obsolete some test options Makefile.am | 2 +- axfer/Makefile.am | 61 +++ axfer/container-au.c | 207 ++++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 581 ++++++++++++++++++++++ axfer/container-voc.c | 843 ++++++++++++++++++++++++++++++++ axfer/container.c | 470 ++++++++++++++++++ axfer/container.h | 126 +++++ axfer/key-event.c | 121 +++++ axfer/key-event.h | 19 + axfer/main.c | 194 ++++++++ axfer/mapper-multiple.c | 271 +++++++++++ axfer/mapper-single.c | 195 ++++++++ axfer/mapper.c | 152 ++++++ axfer/mapper.h | 87 ++++ axfer/misc.h | 16 + axfer/options.c | 980 ++++++++++++++++++++++++++++++++++++++ axfer/options.h | 92 ++++ axfer/subcmd-list.c | 234 +++++++++ axfer/subcmd-transfer.c | 525 ++++++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 39 ++ axfer/test/container-test.c | 299 ++++++++++++ axfer/test/generator.c | 260 ++++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 +++++++++++++++++++ axfer/test/vumeter-test.c | 184 +++++++ axfer/vumeter.c | 546 +++++++++++++++++++++ axfer/vumeter.h | 55 +++ axfer/waiter-epoll.c | 81 ++++ axfer/waiter-poll.c | 68 +++ axfer/waiter.c | 84 ++++ axfer/waiter.h | 52 ++ axfer/xfer-libasound-irq-mmap.c | 271 +++++++++++ axfer/xfer-libasound-irq-rw.c | 520 ++++++++++++++++++++ axfer/xfer-libasound-timer-mmap.c | 474 ++++++++++++++++++ axfer/xfer-libasound.c | 737 ++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 60 +++ axfer/xfer.c | 191 ++++++++ axfer/xfer.h | 76 +++ configure.ac | 3 +- 41 files changed, 9796 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/key-event.c create mode 100644 axfer/key-event.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/options.c create mode 100644 axfer/options.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/test/vumeter-test.c create mode 100644 axfer/vumeter.c create mode 100644 axfer/vumeter.h create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound-timer-mmap.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h -- 2.11.0

2 40

Re: [alsa-devel] [Patch v6 1/7] slimbus: Device management on SLIMbus
by Rishi Bhattacharya 25 Oct '17

25 Oct '17

1 0

[alsa-devel] [PATCH] ALSA: wavefront: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- include/sound/snd_wavefront.h | 1 + sound/isa/wavefront/wavefront_midi.c | 10 +++++----- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/include/sound/snd_wavefront.h b/include/sound/snd_wavefront.h index cd0bab1ef6f1..fcd770fdcda5 100644 --- a/include/sound/snd_wavefront.h +++ b/include/sound/snd_wavefront.h @@ -28,6 +28,7 @@ struct _snd_wavefront_midi { struct snd_rawmidi_substream *substream_output[2]; struct snd_rawmidi_substream *substream_input[2]; struct timer_list timer; + snd_wavefront_card_t *timer_card; spinlock_t open; spinlock_t virtual; /* protects isvirtual */ }; diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c index 2aa05f3aaa38..556b14738970 100644 --- a/sound/isa/wavefront/wavefront_midi.c +++ b/sound/isa/wavefront/wavefront_midi.c @@ -349,10 +349,10 @@ static void snd_wavefront_midi_input_trigger(struct snd_rawmidi_substream *subst spin_unlock_irqrestore (&midi->virtual, flags); } -static void snd_wavefront_midi_output_timer(unsigned long data) +static void snd_wavefront_midi_output_timer(struct timer_list *t) { - snd_wavefront_card_t *card = (snd_wavefront_card_t *)data; - snd_wavefront_midi_t *midi = &card->wavefront.midi; + snd_wavefront_midi_t *midi = from_timer(midi, t, timer); + snd_wavefront_card_t *card = midi->timer_card; unsigned long flags; spin_lock_irqsave (&midi->virtual, flags); @@ -383,9 +383,9 @@ static void snd_wavefront_midi_output_trigger(struct snd_rawmidi_substream *subs if (up) { if ((midi->mode[mpu] & MPU401_MODE_OUTPUT_TRIGGER) == 0) { if (!midi->istimer) { - setup_timer(&midi->timer, + timer_setup(&midi->timer, snd_wavefront_midi_output_timer, - (unsigned long) substream->rmidi->card->private_data); + 0); mod_timer(&midi->timer, 1 + jiffies); } midi->istimer++; -- 2.7.4 -- Kees Cook Pixel Security

2 2

[alsa-devel] snd_pcm_ioplug_avail_update() confusion
by Rob Duncan 24 Oct '17

24 Oct '17

I'm working on an external IO plugin based on ioplug and I'm confused about the behavior of snd_pcm_ioplug_avail_update(). Specifically this call (src/pcm/pcm_ioplug.c:657): result = io->data->callback->transfer(io->data, areas, offset, size); Why isn't there a subsequent call to snd_pcm_mmap_appl_forward() like there is in ioplug_priv_transfer_areas()? It seems that without such a call the captured data will be lost, or have I missed something? Thanks, Rob.

2 1

[alsa-devel] [PATCH] ALSA: sb: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- include/sound/sb.h | 1 + sound/isa/sb/emu8000_pcm.c | 6 +++--- sound/isa/sb/sb8_midi.c | 12 ++++++------ 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/include/sound/sb.h b/include/sound/sb.h index bacefaee411a..4aef395189ec 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -83,6 +83,7 @@ struct snd_sb { unsigned int playback_format; unsigned int capture_format; struct timer_list midi_timer; + struct snd_rawmidi_substream *timer_substream; unsigned int p_dma_size; unsigned int p_period_size; unsigned int c_dma_size; diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 8f34551abd8d..bc5af71d3bdb 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -193,9 +193,9 @@ static inline int emu8k_get_curpos(struct snd_emu8k_pcm *rec, int ch) * timer interrupt handler * check the current position and update the period if necessary. */ -static void emu8k_pcm_timer_func(unsigned long data) +static void emu8k_pcm_timer_func(struct timer_list *t) { - struct snd_emu8k_pcm *rec = (struct snd_emu8k_pcm *)data; + struct snd_emu8k_pcm *rec = from_timer(rec, t, timer); int ptr, delta; spin_lock(&rec->timer_lock); @@ -241,7 +241,7 @@ static int emu8k_pcm_open(struct snd_pcm_substream *subs) runtime->private_data = rec; spin_lock_init(&rec->timer_lock); - setup_timer(&rec->timer, emu8k_pcm_timer_func, (unsigned long)rec); + timer_setup(&rec->timer, emu8k_pcm_timer_func, 0); runtime->hw = emu8k_pcm_hw; runtime->hw.buffer_bytes_max = emu->mem_size - LOOP_BLANK_SIZE * 3; diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c index bd672abb4854..cf23dc5d0507 100644 --- a/sound/isa/sb/sb8_midi.c +++ b/sound/isa/sb/sb8_midi.c @@ -209,10 +209,10 @@ static void snd_sb8dsp_midi_output_write(struct snd_rawmidi_substream *substream } } -static void snd_sb8dsp_midi_output_timer(unsigned long data) +static void snd_sb8dsp_midi_output_timer(struct timer_list *t) { - struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *) data; - struct snd_sb * chip = substream->rmidi->private_data; + struct snd_sb *chip = from_timer(chip, t, midi_timer); + struct snd_rawmidi_substream *substream = chip->timer_substream; unsigned long flags; spin_lock_irqsave(&chip->open_lock, flags); @@ -230,9 +230,9 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre spin_lock_irqsave(&chip->open_lock, flags); if (up) { if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) { - setup_timer(&chip->midi_timer, - snd_sb8dsp_midi_output_timer, - (unsigned long) substream); + chip->timer_substream = substream; + timer_setup(&chip->midi_timer, + snd_sb8dsp_midi_output_timer, 0); mod_timer(&chip->midi_timer, 1 + jiffies); chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER; } -- 2.7.4 -- Kees Cook Pixel Security

2 2

[alsa-devel] [PATCH] ALSA: emux: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/synth/emux/emux.c | 2 +- sound/synth/emux/emux_synth.c | 4 ++-- sound/synth/emux/emux_voice.h | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index b9981e8c0027..b840ff2dcfbb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -53,7 +53,7 @@ int snd_emux_new(struct snd_emux **remu) emu->max_voices = 0; emu->use_time = 0; - setup_timer(&emu->tlist, snd_emux_timer_callback, (unsigned long)emu); + timer_setup(&emu->tlist, snd_emux_timer_callback, 0); emu->timer_active = 0; *remu = emu; diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 599551b5af44..9fa696b0dbde 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -202,9 +202,9 @@ snd_emux_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) * * release the pending note-offs */ -void snd_emux_timer_callback(unsigned long data) +void snd_emux_timer_callback(struct timer_list *t) { - struct snd_emux *emu = (struct snd_emux *) data; + struct snd_emux *emu = from_timer(emu, t, tlist); struct snd_emux_voice *vp; unsigned long flags; int ch, do_again = 0; diff --git a/sound/synth/emux/emux_voice.h b/sound/synth/emux/emux_voice.h index a7073c371bcc..326fa8993d7b 100644 --- a/sound/synth/emux/emux_voice.h +++ b/sound/synth/emux/emux_voice.h @@ -55,7 +55,7 @@ void snd_emux_update_channel(struct snd_emux_port *port, struct snd_midi_channel *chan, int update); void snd_emux_update_port(struct snd_emux_port *port, int update); -void snd_emux_timer_callback(unsigned long data); +void snd_emux_timer_callback(struct timer_list *t); /* emux_effect.c */ #ifdef SNDRV_EMUX_USE_RAW_EFFECT -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ALSA: drivers: Convert timers to use timer_setup()
by Kees Cook 24 Oct '17

24 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Markus Elfring <elfring(a)users.sourceforge.net> Cc: David Howells <dhowells(a)redhat.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/drivers/aloop.c | 7 +++---- sound/drivers/dummy.c | 7 +++---- sound/drivers/mpu401/mpu401_uart.c | 7 +++---- sound/drivers/mtpav.c | 7 +++---- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/drivers/opl3/opl3_seq.c | 2 +- sound/drivers/opl3/opl3_voice.h | 2 +- sound/drivers/serial-u16550.c | 7 +++---- 8 files changed, 19 insertions(+), 24 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 135adb17703c..afac886ffa28 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -529,9 +529,9 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) return running; } -static void loopback_timer_function(unsigned long data) +static void loopback_timer_function(struct timer_list *t) { - struct loopback_pcm *dpcm = (struct loopback_pcm *)data; + struct loopback_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; spin_lock_irqsave(&dpcm->cable->lock, flags); @@ -675,8 +675,7 @@ static int loopback_open(struct snd_pcm_substream *substream) } dpcm->loopback = loopback; dpcm->substream = substream; - setup_timer(&dpcm->timer, loopback_timer_function, - (unsigned long)dpcm); + timer_setup(&dpcm->timer, loopback_timer_function, 0); cable = loopback->cables[substream->number][dev]; if (!cable) { diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index c0939a0164a6..7b2b1f766b00 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -306,9 +306,9 @@ static int dummy_systimer_prepare(struct snd_pcm_substream *substream) return 0; } -static void dummy_systimer_callback(unsigned long data) +static void dummy_systimer_callback(struct timer_list *t) { - struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data; + struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; int elapsed = 0; @@ -343,8 +343,7 @@ static int dummy_systimer_create(struct snd_pcm_substream *substream) if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - setup_timer(&dpcm->timer, dummy_systimer_callback, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, dummy_systimer_callback, 0); spin_lock_init(&dpcm->lock); dpcm->substream = substream; return 0; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index b997222274bd..3e745f47dd2f 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -169,9 +169,9 @@ EXPORT_SYMBOL(snd_mpu401_uart_interrupt_tx); * timer callback * reprogram the timer and call the interrupt job */ -static void snd_mpu401_uart_timer(unsigned long data) +static void snd_mpu401_uart_timer(struct timer_list *t) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *)data; + struct snd_mpu401 *mpu = from_timer(mpu, t, timer); unsigned long flags; spin_lock_irqsave(&mpu->timer_lock, flags); @@ -191,8 +191,7 @@ static void snd_mpu401_uart_add_timer (struct snd_mpu401 *mpu, int input) spin_lock_irqsave (&mpu->timer_lock, flags); if (mpu->timer_invoked == 0) { - setup_timer(&mpu->timer, snd_mpu401_uart_timer, - (unsigned long)mpu); + timer_setup(&mpu->timer, snd_mpu401_uart_timer, 0); mod_timer(&mpu->timer, 1 + jiffies); } mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER : diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 0f6392001e30..547662e02fcc 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -406,10 +406,10 @@ static void snd_mtpav_input_trigger(struct snd_rawmidi_substream *substream, int * timer interrupt for outputs */ -static void snd_mtpav_output_timer(unsigned long data) +static void snd_mtpav_output_timer(struct timer_list *t) { unsigned long flags; - struct mtpav *chip = (struct mtpav *)data; + struct mtpav *chip = from_timer(chip, t, timer); int p; spin_lock_irqsave(&chip->spinlock, flags); @@ -707,8 +707,7 @@ static int snd_mtpav_probe(struct platform_device *dev) mtp_card->share_irq = 0; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; - setup_timer(&mtp_card->timer, snd_mtpav_output_timer, - (unsigned long) mtp_card); + timer_setup(&mtp_card->timer, snd_mtpav_output_timer, 0); card->private_free = snd_mtpav_free; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 13c0a7e1bc2b..bb3f3a5a6951 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -238,10 +238,10 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, /* * System timer interrupt function */ -void snd_opl3_timer_func(unsigned long data) +void snd_opl3_timer_func(struct timer_list *t) { - struct snd_opl3 *opl3 = (struct snd_opl3 *)data; + struct snd_opl3 *opl3 = from_timer(opl3, t, tlist); unsigned long flags; int again = 0; int i; diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index d3e91be8b23a..5f881c4cdf1f 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -248,7 +248,7 @@ static int snd_opl3_seq_probe(struct device *_dev) } /* setup system timer */ - setup_timer(&opl3->tlist, snd_opl3_timer_func, (unsigned long) opl3); + timer_setup(&opl3->tlist, snd_opl3_timer_func, 0); spin_lock_init(&opl3->sys_timer_lock); opl3->sys_timer_status = 0; diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h index eaef435e0528..a2445163008e 100644 --- a/sound/drivers/opl3/opl3_voice.h +++ b/sound/drivers/opl3/opl3_voice.h @@ -37,7 +37,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_chann void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset); void snd_opl3_calc_volume(unsigned char *reg, int vel, struct snd_midi_channel *chan); -void snd_opl3_timer_func(unsigned long data); +void snd_opl3_timer_func(struct timer_list *t); /* Prototypes for opl3_drums.c */ void snd_opl3_load_drums(struct snd_opl3 *opl3); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 88e66ea0306d..0a67b8b9f176 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -309,12 +309,12 @@ static irqreturn_t snd_uart16550_interrupt(int irq, void *dev_id) } /* When the polling mode, this function calls snd_uart16550_io_loop. */ -static void snd_uart16550_buffer_timer(unsigned long data) +static void snd_uart16550_buffer_timer(struct timer_list *t) { unsigned long flags; struct snd_uart16550 *uart; - uart = (struct snd_uart16550 *)data; + uart = from_timer(uart, t, buffer_timer); spin_lock_irqsave(&uart->open_lock, flags); snd_uart16550_del_timer(uart); snd_uart16550_io_loop(uart); @@ -828,8 +828,7 @@ static int snd_uart16550_create(struct snd_card *card, uart->prev_in = 0; uart->rstatus = 0; memset(uart->prev_status, 0x80, sizeof(unsigned char) * SNDRV_SERIAL_MAX_OUTS); - setup_timer(&uart->buffer_timer, snd_uart16550_buffer_timer, - (unsigned long)uart); + timer_setup(&uart->buffer_timer, snd_uart16550_buffer_timer, 0); uart->timer_running = 0; /* Register device */ -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ALSA: hda controller - handle -1 value as error in snd_hdac_bus_parse_capabilities()
by Jaroslav Kysela 24 Oct '17

24 Oct '17

We have seen this value on a development platform. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/hda/hdac_controller.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc1801b3a..4a1823290b2d 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -278,6 +278,11 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) do { cur_cap = _snd_hdac_chip_readl(bus, offset); + if (cur_cap == 0xffffffff) { + dev_dbg(bus->dev, "Wrong capability -1\n"); + break; + } + dev_dbg(bus->dev, "Capability version: 0x%x\n", (cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF); -- 2.13.6

2 1

[alsa-devel] conflict between alsa-lib and libasound2-data regarding alsa.conf file:
by arne 24 Oct '17

24 Oct '17

Hi, Debian conflicts between alsa-lib_1.1.4 and libasound2-data 1.1.3: in the source tree of alsa-lib I do: ./configure && make && sudo checkinstall and the result is: Unpacking alsa-lib (1.1.4.1-1) ... dpkg: error processing archive /home/temp/alsa-lib-1.1.4.1/alsa-lib_1.1.4.1-1_amd64.deb (--install): trying to overwrite '/usr/share/alsa/alsa.conf', which is also in package libasound2-data 1.1.3-5 This is on a Debian amd64 stable computer Thanks!

3 2

[alsa-devel] New to this list. Proposed patch to soc-utils.c
by Albin Stigö 24 Oct '17

24 Oct '17

Hello, I'm new to this list so please excuse my ignorance. I'm an anesthesiologist by profession so I might be a bit out of my depth here... I was doing some experimentation with custom i2s codecs (FPGAs) using the snd-soc-dummy codec (defined in soc-utils.c). I found this codec to be needlessly restrictive with sample rates allowed (SNDRV_PCM_RATE_8000_192000). I needed to use more "exotic" sample rates so I came up with a patch. I purpose using SNDRV_PCM_RATE_KNOT and setting some limits: .rate_min = 8000, .rate_max = 384000, It's my understanding that SNDRV_PCM_RATE_CONTINIOUS is used for "known" rates and SNDRV_PCM_RATE_KNOT for more or less arbitrary rates? I'm willing to submit my patch but might need some mentoring in the process. I don't think this simple patch could break anything..? --Albin

1 1

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Subhransu S. Prusty 24 Oct '17

24 Oct '17

For certain platforms we require to start the clocks (mclk/sclk/fs) before the stream start. Example: for few systems, codec needs the mclk/sclk/fs to be enabled early for a successful clock synchronization and for some others, clock need to be enabled at boot and should be ON always. By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Skylake driver creates a platform driver with the clock information and register the clk ops callback. kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. set_dma_control API can be used with different payload configuration. Modify the arguments to take configurable parameters. Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Jaikrishna Nemallapudi (5): ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments ASoC: Intel: Skylake: Parse nhlt to populate clock information ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock ASoC: Intel: Skylake: Register clock device and ops ASoC: Intel: Skylake: Add ssp clock driver sound/soc/intel/Kconfig | 8 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 94 ++++++++- sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 56 ++++++ sound/soc/intel/skylake/skl-messages.c | 109 +++++++++- sound/soc/intel/skylake/skl-nhlt.c | 148 ++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.c | 287 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 124 ++++++++++++ sound/soc/intel/skylake/skl-topology.h | 4 +- sound/soc/intel/skylake/skl.c | 235 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 12 ++ 11 files changed, 1070 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-i2s.h create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h -- 1.9.1

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[alsa-devel] ASoC: writing basic custom codecs?
by Albin Stigö 24 Oct '17

24 Oct '17

Hello, I have implemented a custom i2s codec on an FPGA. Because of how the FPGA is clocked I'm getting samples at 39ksps, which to my surprise worked great after some simple kernel patching (the codec is bit & frame master). For experimentation I used the snd-soc-dummy coded defined in soc-utils.c. Now I'm trying to write a very simple codec to "lock" the rate to 39000: https://github.com/ast/vfzfpga-codec The driver loads just fine but I just get "not registered" when I try to use the codec. I feel I'm missing something very obvious here. I've been through the documentation and tried to read the sources but I'm a bit stuck.. What am I missing? --Albin

1 0

[alsa-devel] Prague Audio miniconference - draft schedule
by Mark Brown 24 Oct '17

24 Oct '17

As previously announced[1] by Takashi our annual Linux audio miniconference will be held this year at the SuSE offices in Prague on Friday 27th October (the week of ELC-E). Thanks again to SuSE for sponsoring this. Following the earlier discussion I've pulled together a list of topics. I've put people's names against everything, mostly the people who suggested things but sometimes also other people who should probably be involved. I'm proposing that we work through things in roughly this order. Does this make sense to people? Any other topics? Summary - API status summary (tiwai) - Userspace release schedule - Docs - Dropping old drivers - Treble overview (Eric) - aplay rewrite (Sakamoto-san) ABI issues - USB PCM handling (tiwai) - mmap no IRQ (Eric) - TLV API issues (Sakamoto-san) - Obsolete dimen from struct snd_ctl_elem_info (Sakamoto-san) - Y2038 (tiwai) Future stuff/internals - Clock handling (Peter) - CCF, multi-clock selection - Digital domains (Charles/Lars/Morimoto-san) - Large regmaps (Charles) [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124623.html

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic problem for Dell machines with alc236
by Hui Wang 24 Oct '17

24 Oct '17

We have several Dell laptops which use the codec alc236, the headset mic can't work on these machines. Following the commit 736f20a70, we add the pin cfg table to make the headset mic work. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701eb7d..02c79d6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6429,6 +6429,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x1b, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60140}, + {0x14, 0x90170110}, + {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60140}, + {0x14, 0x90170150}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x02211020}), -- 2.7.4

2 1

[alsa-devel] [PATCH v2 34/37] ALSA/dummy: Replace tasklet with softirq hrtimer
by Anna-Maria Gleixner 24 Oct '17

24 Oct '17

From: Thomas Gleixner <tglx(a)linutronix.de> The tasklet is used to defer the execution of snd_pcm_period_elapsed() to the softirq context. Using the HRTIMER_MODE_SOFT mode invokes the timer callback in softirq context as well which renders the tasklet useless. [o-takashi: avoid stall due to a call of hrtimer_cancel() on a callback of hrtimer] Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Anna-Maria Gleixner <anna-maria(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/20170905161820.jtysvxtfleunbbmf@breakpoint.cc --- On 2017-09-06 01:05:43 [+0900], Takashi Sakamoto wrote: > As Iwai-san mentioned, in this function, .trigger can be called in two > cases; XRUN occurs and draining is done. Thus, let you change the comment as > 'In cases of XRUN and draining, this calls .trigger to stop PCM substream.'. > I'm sorry to trouble you. > > snd_pcm_period_elapsed() > ->snd_pcm_update_hw_ptr0() > ->snd_pcm_update_state() > ->snd_pcm_drain_done() > ... > ->.trigger(TRIGGER_STOP) > ->xrun() > ->snd_pcm_stop() > ... > ->.trigger(TRIGGER_STOP) > I think you asked me just to update the comment. Did I do it right? v2...v3: updated the comment as per Takashi Sakamoto's suggestion. v1...v2: merged Takashi Sakamoto fixup of the original patch into v2. sound/drivers/dummy.c | 27 ++++++++++++--------------- 1 file changed, 12 insertions(+), 15 deletions(-) --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -376,17 +376,9 @@ struct dummy_hrtimer_pcm { ktime_t period_time; atomic_t running; struct hrtimer timer; - struct tasklet_struct tasklet; struct snd_pcm_substream *substream; }; -static void dummy_hrtimer_pcm_elapsed(unsigned long priv) -{ - struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; - if (atomic_read(&dpcm->running)) - snd_pcm_period_elapsed(dpcm->substream); -} - static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) { struct dummy_hrtimer_pcm *dpcm; @@ -394,7 +386,14 @@ static enum hrtimer_restart dummy_hrtime dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); if (!atomic_read(&dpcm->running)) return HRTIMER_NORESTART; - tasklet_schedule(&dpcm->tasklet); + /* + * In cases of XRUN and draining, this calls .trigger to stop PCM + * substream. + */ + snd_pcm_period_elapsed(dpcm->substream); + if (!atomic_read(&dpcm->running)) + return HRTIMER_NORESTART; + hrtimer_forward_now(timer, dpcm->period_time); return HRTIMER_RESTART; } @@ -404,7 +403,7 @@ static int dummy_hrtimer_start(struct sn struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; dpcm->base_time = hrtimer_cb_get_time(&dpcm->timer); - hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL); + hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL_SOFT); atomic_set(&dpcm->running, 1); return 0; } @@ -414,14 +413,14 @@ static int dummy_hrtimer_stop(struct snd struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; atomic_set(&dpcm->running, 0); - hrtimer_cancel(&dpcm->timer); + if (!hrtimer_callback_running(&dpcm->timer)) + hrtimer_cancel(&dpcm->timer); return 0; } static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { hrtimer_cancel(&dpcm->timer); - tasklet_kill(&dpcm->tasklet); } static snd_pcm_uframes_t @@ -466,12 +465,10 @@ static int dummy_hrtimer_create(struct s if (!dpcm) return -ENOMEM; substream->runtime->private_data = dpcm; - hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL_SOFT); dpcm->timer.function = dummy_hrtimer_callback; dpcm->substream = substream; atomic_set(&dpcm->running, 0); - tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, - (unsigned long)dpcm); return 0; }

2 1

[alsa-devel] [PATCH v3] ALSA: hda: Abort capability probe on invalid capability
by Rakesh Ughreja 24 Oct '17

24 Oct '17

On reading wrong capability pointer values driver may crash, so whenever driver discovers unknown HDA capability, log it as error and stop traversing the link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

3 2

[alsa-devel] how to dynamic link one cpu dai to codec dai?
by Gou, Kylin 24 Oct '17

24 Oct '17

Hi Sir, I registered 2 codec dai and 1 cpu dai . However , I want to link cpu dai to different codec dai , can we dynamic link cpu dai to each codec dai? I didn't find any document for it. Thanks, Kylin ******************************* IMPORTANT MESSAGE ****************************** This transmission may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the information contained herein (including any reliance thereon) is STRICTLY PROHIBITED. If you received this transmission in error, please immediately contact the sender and destroy the material in its entirety, whether in electronic or hard copy format. Thank you. ********************************************************************************

1 0

[alsa-devel] Prague Audio miniconference - topics for discussion?
by Mark Brown 23 Oct '17

23 Oct '17

As previously announced[1] by Takashi our annual Linux audio miniconference will be held this year at the SuSE offices in Prague on 27th October (the week of ELC-E). Thanks again to SuSe for sponsoring this. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! I'll start things off by mentining the TLV issues that Sakamoto-san was raising - do we have any better ideas to handle larger binary controls than what's currently being done? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124623.html

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[alsa-devel] [PATCH] ASoC: Intel: Add depends on X86
by Vinod Koul 23 Oct '17

23 Oct '17

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Make all Intel audio drivers dependent on X86 to avoid compilation errors for s390 and xtensa architectures. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 1dcc9e456610..a59dda39007e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" + depends on X86 select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5651: Convert rt5651 micbias1 to a supply widget
by Carlo Caione 23 Oct '17

23 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> MICBIAS widget type has been deprecated. Convert it to a SUPPLY widget. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index da60b28ba3df..28f7210cec91 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -883,8 +883,8 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { /* micbias */ SND_SOC_DAPM_SUPPLY("LDO", RT5651_PWR_ANLG1, RT5651_PWR_LDO_BIT, 0, NULL, 0), - SND_SOC_DAPM_MICBIAS("micbias1", RT5651_PWR_ANLG2, - RT5651_PWR_MB1_BIT, 0), + SND_SOC_DAPM_SUPPLY("micbias1", RT5651_PWR_ANLG2, + RT5651_PWR_MB1_BIT, 0, NULL, 0), /* Input Lines */ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), -- 2.14.2

3 3

[alsa-devel] [asoc:topic/intel 38/38] sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant
by kbuild test robot 23 Oct '17

23 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 commit: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 [38/38] ASoC: Intel: clarify Kconfig dependencies config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout f6a118a800e35af2c63f90cbcc23093f4b53b3a2 # save the attached .config to linux build tree make.cross ARCH=xtensa All errors (new ones prefixed by >>): >> sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant MODULE_DESCRIPTION("Intel SST Firmware Loader"); ^ sound/soc/intel/common/sst-firmware.c:1279:16: error: expected declaration specifiers or '...' before string constant MODULE_LICENSE("GPL v2"); ^ -- sound/soc/intel/haswell/sst-haswell-pcm.c: In function 'create_adsp_page_table': >> sound/soc/intel/haswell/sst-haswell-pcm.c:446:2: error: implicit declaration of function 'snd_sgbuf_aligned_pages' [-Werror=implicit-function-declaration] pages = snd_sgbuf_aligned_pages(size); ^ cc1: some warnings being treated as errors vim +1278 sound/soc/intel/common/sst-firmware.c a395bdd6 Takashi Iwai 2016-07-11 1277 a395bdd6 Takashi Iwai 2016-07-11 @1278 MODULE_DESCRIPTION("Intel SST Firmware Loader"); :::::: The code at line 1278 was first introduced by commit :::::: a395bdd6b24b692adbce0df6510ec9f2af57573e ASoC: intel: Fix sst-dsp dependency on dw stuff :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] HDMI LPE audio on Intel Compute Stick 2nd Gen (CHT)
by Ian W MORRISON 22 Oct '17

22 Oct '17

Hi Pierre, I've just noticed that non of the kernels from v4.13 and up run on my Intel Compute Stick (2nd gen Cherry Trail STK1AW32SC) unit and have HDMI LPE audio working. Before I spend anytime looking into this can you confirm if you have the same issue as I just want to rule out a hardware malfunction with my unit first? Thanks, Ian

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[alsa-devel] [asoc:topic/intel 38/38] sound/soc//intel/skylake/skl-sst-cldma.c:92:22: error: implicit declaration of function 'virt_to_phys'
by kbuild test robot 21 Oct '17

21 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 commit: f6a118a800e35af2c63f90cbcc23093f4b53b3a2 [38/38] ASoC: Intel: clarify Kconfig dependencies config: s390-allyesconfig (attached as .config) compiler: s390x-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout f6a118a800e35af2c63f90cbcc23093f4b53b3a2 # save the attached .config to linux build tree make.cross ARCH=s390 All errors (new ones prefixed by >>): sound/soc//intel/skylake/skl-sst-cldma.c: In function 'skl_cldma_setup_bdle': >> sound/soc//intel/skylake/skl-sst-cldma.c:92:22: error: implicit declaration of function 'virt_to_phys' [-Werror=implicit-function-declaration] phys_addr_t addr = virt_to_phys(dmab_data->area + ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/virt_to_phys +92 sound/soc//intel/skylake/skl-sst-cldma.c a4386450b Jeeja KP 2015-12-18 82 914426c86 Vinod Koul 2015-07-10 83 /* Code loader helper APIs */ 914426c86 Vinod Koul 2015-07-10 84 static void skl_cldma_setup_bdle(struct sst_dsp *ctx, 914426c86 Vinod Koul 2015-07-10 85 struct snd_dma_buffer *dmab_data, 914426c86 Vinod Koul 2015-07-10 86 u32 **bdlp, int size, int with_ioc) 914426c86 Vinod Koul 2015-07-10 87 { 914426c86 Vinod Koul 2015-07-10 88 u32 *bdl = *bdlp; 914426c86 Vinod Koul 2015-07-10 89 914426c86 Vinod Koul 2015-07-10 90 ctx->cl_dev.frags = 0; 914426c86 Vinod Koul 2015-07-10 91 while (size > 0) { 914426c86 Vinod Koul 2015-07-10 @92 phys_addr_t addr = virt_to_phys(dmab_data->area + 914426c86 Vinod Koul 2015-07-10 93 (ctx->cl_dev.frags * ctx->cl_dev.bufsize)); 914426c86 Vinod Koul 2015-07-10 94 914426c86 Vinod Koul 2015-07-10 95 bdl[0] = cpu_to_le32(lower_32_bits(addr)); 914426c86 Vinod Koul 2015-07-10 96 bdl[1] = cpu_to_le32(upper_32_bits(addr)); 914426c86 Vinod Koul 2015-07-10 97 914426c86 Vinod Koul 2015-07-10 98 bdl[2] = cpu_to_le32(ctx->cl_dev.bufsize); 914426c86 Vinod Koul 2015-07-10 99 914426c86 Vinod Koul 2015-07-10 100 size -= ctx->cl_dev.bufsize; 914426c86 Vinod Koul 2015-07-10 101 bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); 914426c86 Vinod Koul 2015-07-10 102 914426c86 Vinod Koul 2015-07-10 103 bdl += 4; 914426c86 Vinod Koul 2015-07-10 104 ctx->cl_dev.frags++; 914426c86 Vinod Koul 2015-07-10 105 } 914426c86 Vinod Koul 2015-07-10 106 } 914426c86 Vinod Koul 2015-07-10 107 :::::: The code at line 92 was first introduced by commit :::::: 914426c8657c1e934b015ffa32e7bfd0cb34f47f ASoC: Intel: Skylake: add code loader DMA operations :::::: TO: Vinod Koul <vinod.koul(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/stm32 5/7] sound/soc/stm/stm32_sai_sub.c:467:7: warning: 'cr1' may be used uninitialized in this function
by kbuild test robot 21 Oct '17

21 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/stm32 head: 5914d285f6b782892a91d6621723fdc41a775b15 commit: 61fb4ff70377cd2c49a3487bdb0156eba6930072 [5/7] ASoC: stm32: sai: Move static settings to DAI init config: xtensa-allyesconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 61fb4ff70377cd2c49a3487bdb0156eba6930072 # save the attached .config to linux build tree make.cross ARCH=xtensa Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_hw_params': >> sound/soc/stm/stm32_sai_sub.c:467:7: warning: 'cr1' may be used uninitialized in this function [-Wmaybe-uninitialized] cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); ^ sound/soc/stm/stm32_sai_sub.c:451:6: note: 'cr1' was declared here int cr1, cr1_mask, ret; ^ vim +/cr1 +467 sound/soc/stm/stm32_sai_sub.c 3e086edf olivier moysan 2017-04-10 445 3e086edf olivier moysan 2017-04-10 446 static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, 3e086edf olivier moysan 2017-04-10 447 struct snd_pcm_substream *substream, 3e086edf olivier moysan 2017-04-10 448 struct snd_pcm_hw_params *params) 3e086edf olivier moysan 2017-04-10 449 { 3e086edf olivier moysan 2017-04-10 450 struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); 3e086edf olivier moysan 2017-04-10 451 int cr1, cr1_mask, ret; 3e086edf olivier moysan 2017-04-10 452 a4529d2b Olivier Moysan 2017-10-19 453 /* a4529d2b Olivier Moysan 2017-10-19 454 * DMA bursts increment is set to 4 words. a4529d2b Olivier Moysan 2017-10-19 455 * SAI fifo threshold is set to half fifo, to keep enough space a4529d2b Olivier Moysan 2017-10-19 456 * for DMA incoming bursts. a4529d2b Olivier Moysan 2017-10-19 457 */ 3e086edf olivier moysan 2017-04-10 458 regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, 3e086edf olivier moysan 2017-04-10 459 SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, a4529d2b Olivier Moysan 2017-10-19 460 SAI_XCR2_FFLUSH | a4529d2b Olivier Moysan 2017-10-19 461 SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); 3e086edf olivier moysan 2017-04-10 462 3e086edf olivier moysan 2017-04-10 463 /* Mode, data format and channel config */ 61fb4ff7 Olivier Moysan 2017-10-19 464 cr1_mask = SAI_XCR1_DS_MASK; 3e086edf olivier moysan 2017-04-10 465 switch (params_format(params)) { 3e086edf olivier moysan 2017-04-10 466 case SNDRV_PCM_FORMAT_S8: 3e086edf olivier moysan 2017-04-10 @467 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); 3e086edf olivier moysan 2017-04-10 468 break; 3e086edf olivier moysan 2017-04-10 469 case SNDRV_PCM_FORMAT_S16_LE: 3e086edf olivier moysan 2017-04-10 470 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_16); 3e086edf olivier moysan 2017-04-10 471 break; 3e086edf olivier moysan 2017-04-10 472 case SNDRV_PCM_FORMAT_S32_LE: 3e086edf olivier moysan 2017-04-10 473 cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_32); 3e086edf olivier moysan 2017-04-10 474 break; 3e086edf olivier moysan 2017-04-10 475 default: 3e086edf olivier moysan 2017-04-10 476 dev_err(cpu_dai->dev, "Data format not supported"); 3e086edf olivier moysan 2017-04-10 477 return -EINVAL; 3e086edf olivier moysan 2017-04-10 478 } 3e086edf olivier moysan 2017-04-10 479 3e086edf olivier moysan 2017-04-10 480 cr1_mask |= SAI_XCR1_MONO; 3e086edf olivier moysan 2017-04-10 481 if ((sai->slots == 2) && (params_channels(params) == 1)) 3e086edf olivier moysan 2017-04-10 482 cr1 |= SAI_XCR1_MONO; 3e086edf olivier moysan 2017-04-10 483 3e086edf olivier moysan 2017-04-10 484 ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); 3e086edf olivier moysan 2017-04-10 485 if (ret < 0) { 3e086edf olivier moysan 2017-04-10 486 dev_err(cpu_dai->dev, "Failed to update CR1 register\n"); 3e086edf olivier moysan 2017-04-10 487 return ret; 3e086edf olivier moysan 2017-04-10 488 } 3e086edf olivier moysan 2017-04-10 489 3e086edf olivier moysan 2017-04-10 490 return 0; 3e086edf olivier moysan 2017-04-10 491 } 3e086edf olivier moysan 2017-04-10 492 :::::: The code at line 467 was first introduced by commit :::::: 3e086edfe0c73daaabd929b926bbe26536272d9a ASoC: stm32: add SAI driver :::::: TO: olivier moysan <olivier.moysan(a)st.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Applied "ASoC: stm32: sai: Fix DMA burst size" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: stm32: sai: Fix DMA burst size has been applied to the asoc tree at git://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 a4529d2b885747bb9393d507cd28649052d733f3 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Thu, 19 Oct 2017 15:03:19 +0200 Subject: [PATCH] ASoC: stm32: sai: Fix DMA burst size Set best burst size tradeoff for 8, 16, 32 bits transfers. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 5e66bf310351..dde8ddf9e777 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -446,12 +446,16 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); int cr1, cr1_mask, ret; - int fth = STM_SAI_FIFO_TH_HALF; - /* FIFO config */ + /* + * DMA bursts increment is set to 4 words. + * SAI fifo threshold is set to half fifo, to keep enough space + * for DMA incoming bursts. + */ regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_SET(fth)); + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* Mode, data format and channel config */ cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); @@ -485,10 +489,6 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, return ret; } - /* DMA config */ - sai->dma_params.maxburst = STM_SAI_FIFO_SIZE * fth / sizeof(u32); - snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&sai->dma_params); - return 0; } @@ -731,7 +731,12 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); - sai->dma_params.maxburst = 1; + /* + * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice, + * as it allows bytes, half-word and words transfers. (See DMA fifos + * constraints). + */ + sai->dma_params.maxburst = 4; /* Buswidth will be set by framework at runtime */ sai->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; -- 2.14.1

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[alsa-devel] Applied "ASoC: stm32: sai: Move static settings to DAI init" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: stm32: sai: Move static settings to DAI init has been applied to the asoc tree at git://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 61fb4ff70377cd2c49a3487bdb0156eba6930072 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Thu, 19 Oct 2017 15:03:18 +0200 Subject: [PATCH] ASoC: stm32: sai: Move static settings to DAI init Audio interface direction and protocol settings does not change at runtime. So, these settings are moved from hw_params function to dai_probe and set_fmt. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dde8ddf9e777..815ef1036199 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -308,12 +308,15 @@ static int stm32_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); - int cr1 = 0, frcr = 0; - int cr1_mask = 0, frcr_mask = 0; + int cr1, frcr = 0; + int cr1_mask, frcr_mask = 0; int ret; dev_dbg(cpu_dai->dev, "fmt %x\n", fmt); + cr1_mask = SAI_XCR1_PRTCFG_MASK; + cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { /* SCK active high for all protocols */ case SND_SOC_DAIFMT_I2S: @@ -340,7 +343,7 @@ static int stm32_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return -EINVAL; } - cr1_mask |= SAI_XCR1_PRTCFG_MASK | SAI_XCR1_CKSTR; + cr1_mask |= SAI_XCR1_CKSTR; frcr_mask |= SAI_XFRCR_FSPOL | SAI_XFRCR_FSOFF | SAI_XFRCR_FSDEF; @@ -458,7 +461,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* Mode, data format and channel config */ - cr1 = SAI_XCR1_PRTCFG_SET(SAI_FREE_PROTOCOL); + cr1_mask = SAI_XCR1_DS_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); @@ -473,11 +476,6 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, dev_err(cpu_dai->dev, "Data format not supported"); return -EINVAL; } - cr1_mask = SAI_XCR1_DS_MASK | SAI_XCR1_PRTCFG_MASK; - - cr1_mask |= SAI_XCR1_RX_TX; - if (STM_SAI_IS_CAPTURE(sai)) - cr1 |= SAI_XCR1_RX_TX; cr1_mask |= SAI_XCR1_MONO; if ((sai->slots == 2) && (params_channels(params) == 1)) @@ -729,6 +727,7 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + int cr1 = 0, cr1_mask; sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX); /* @@ -745,7 +744,11 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); - return 0; + cr1_mask = SAI_XCR1_RX_TX; + if (STM_SAI_IS_CAPTURE(sai)) + cr1 |= SAI_XCR1_RX_TX; + + return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1); } static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = { -- 2.14.1

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[alsa-devel] Applied "ASoC: move ACPI common code out of Intel/sst tree" to the asoc tree
by Mark Brown 21 Oct '17

21 Oct '17

The patch ASoC: move ACPI common code out of Intel/sst tree has been applied to the asoc tree at git://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 7feb2f786a46d34cac760d5e246dfbdafafa0e96 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 12 Oct 2017 18:49:38 -0500 Subject: [PATCH] ASoC: move ACPI common code out of Intel/sst tree ACPI support is not specific to the Intel/SST driver. Move the enumeration and matching code which is not hardware-dependent to sound/soc and rename relevant sst_acpi_ structures and functions with snd_soc_acpi_ prefix soc-acpi.h is protected by a #ifndef __LINUX_SND_SOC_ACPI_H for consistency with all other SoC .h files: grep -L __LINUX include/sound/soc* | wc -l 0 grep __LINUX include/sound/soc* | wc -l 14 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-acpi.h | 100 +++++++++++++++++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 ++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst/sst_acpi.c | 24 ++--- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +-- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +-- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 2 - sound/soc/intel/common/sst-acpi.c | 18 ++-- sound/soc/intel/common/sst-acpi.h | 82 ----------------- sound/soc/intel/skylake/skl.c | 40 ++++----- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 ++++++------ 16 files changed, 203 insertions(+), 172 deletions(-) create mode 100644 include/sound/soc-acpi.h delete mode 100644 sound/soc/intel/common/sst-acpi.h rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h new file mode 100644 index 000000000000..918d78e8c8d6 --- /dev/null +++ b/include/sound/soc-acpi.h @@ -0,0 +1,100 @@ +/* + * Copyright (C) 2013-15, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __LINUX_SND_SOC_ACPI_H +#define __LINUX_SND_SOC_ACPI_H + +#include <linux/stddef.h> +#include <linux/acpi.h> + +struct snd_soc_acpi_package_context { + char *name; /* package name */ + int length; /* number of elements */ + struct acpi_buffer *format; + struct acpi_buffer *state; + bool data_valid; +}; + +#if IS_ENABLED(CONFIG_ACPI) +/* translation fron HID to I2C name, needed for DAI codec_name */ +const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx); +#else +static inline const char * +snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +static inline bool +snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx) +{ + return false; +} +#endif + +/* acpi match */ +struct snd_soc_acpi_mach * +snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); + +/* acpi check hid */ +bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); + +/** + * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are + * related to the hardware, except for the firmware and topology file names. + * + * @id: ACPI ID (usually the codec's) used to find a matching machine driver. + * @drv_name: machine driver name + * @fw_filename: firmware file name. + * @board: board name + * @machine_quirk: pointer to quirk, usually based on DMI information when + * ACPI ID alone is not sufficient, wrong or misleading + * @quirk_data: data used to uniquely identify a machine, usually a list of + * audio codecs whose presence if checked with ACPI + * @pdata: intended for platform data or machine specific-ops. This structure + * is not constant since this field may be updated at run-time + */ +/* Descriptor for SST ASoC machine driver */ +struct snd_soc_acpi_mach { + const u8 id[ACPI_ID_LEN]; + const char *drv_name; + const char *fw_filename; + const char *board; + struct snd_soc_acpi_mach * (*machine_quirk)(void *arg); + const void *quirk_data; + void *pdata; +}; + +#define SND_SOC_ACPI_MAX_CODECS 3 + +/** + * struct snd_soc_acpi_codecs: Structure to hold secondary codec information + * apart from the matched one, this data will be passed to the quirk function + * to match with the ACPI detected devices + * + * @num_codecs: number of secondary codecs used in the platform + * @codecs: holds the codec IDs + * + */ +struct snd_soc_acpi_codecs { + int num_codecs; + u8 codecs[SND_SOC_ACPI_MAX_CODECS][ACPI_ID_LEN]; +}; + +/* check all codecs */ +struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg); + +#endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index c0abad2067e1..d22758165496 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,9 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool +config SND_SOC_ACPI + tristate + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/amd/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 39c27a58158d..95f2a73f9aed 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,12 @@ ifneq ($(CONFIG_SND_SOC_AC97_BUS),) snd-soc-core-objs += soc-ac97.o endif +ifneq ($(CONFIG_SND_SOC_ACPI),) +snd-soc-acpi-objs := soc-acpi.o +endif + +obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3c7f554ec30..d03199cc61e0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_INTEL_SST_ACPI config SND_SOC_INTEL_SST_MATCH tristate + select SND_SOC_ACPI if ACPI config SND_SOC_INTEL_HASWELL tristate diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 4d85ead243f1..8d82bc8dc4ac 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -40,9 +40,9 @@ #include <acpi/acpi_bus.h> #include <asm/cpu_device_id.h> #include <asm/iosf_mbi.h> +#include <sound/soc-acpi.h> #include "../sst-mfld-platform.h" #include "../../common/sst-dsp.h" -#include "../../common/sst-acpi.h" #include "sst.h" /* LPE viewpoint addresses */ @@ -284,7 +284,7 @@ static int sst_acpi_probe(struct platform_device *pdev) int ret = 0; struct intel_sst_drv *ctx; const struct acpi_device_id *id; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct platform_device *mdev; struct platform_device *plat_dev; struct sst_platform_info *pdata; @@ -296,8 +296,8 @@ static int sst_acpi_probe(struct platform_device *pdev) return -ENODEV; dev_dbg(dev, "for %s\n", id->id); - mach = (struct sst_acpi_mach *)id->driver_data; - mach = sst_acpi_find_machine(mach); + mach = (struct snd_soc_acpi_mach *)id->driver_data; + mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(dev, "No matching machine driver found\n"); return -ENODEV; @@ -435,7 +435,7 @@ static const struct dmi_system_id cht_table[] = { }; -static struct sst_acpi_mach cht_surface_mach = { +static struct snd_soc_acpi_mach cht_surface_mach = { .id = "10EC5640", .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", @@ -443,7 +443,7 @@ static struct sst_acpi_mach cht_surface_mach = { .pdata = &chv_platform_data, }; -static struct sst_acpi_mach byt_thinkpad_10 = { +static struct snd_soc_acpi_mach byt_thinkpad_10 = { .id = "10EC5640", .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", @@ -451,9 +451,9 @@ static struct sst_acpi_mach byt_thinkpad_10 = { .pdata = &byt_rvp_platform_data, }; -static struct sst_acpi_mach *cht_quirk(void *arg) +static struct snd_soc_acpi_mach *cht_quirk(void *arg) { - struct sst_acpi_mach *mach = arg; + struct snd_soc_acpi_mach *mach = arg; dmi_check_system(cht_table); @@ -463,9 +463,9 @@ static struct sst_acpi_mach *cht_quirk(void *arg) return mach; } -static struct sst_acpi_mach *byt_quirk(void *arg) +static struct snd_soc_acpi_mach *byt_quirk(void *arg) { - struct sst_acpi_mach *mach = arg; + struct snd_soc_acpi_mach *mach = arg; dmi_check_system(byt_table); @@ -476,7 +476,7 @@ static struct sst_acpi_mach *byt_quirk(void *arg) } -static struct sst_acpi_mach sst_acpi_bytcr[] = { +static struct snd_soc_acpi_mach sst_acpi_bytcr[] = { { .id = "10EC5640", .drv_name = "bytcr_rt5640", @@ -560,7 +560,7 @@ static struct sst_acpi_mach sst_acpi_bytcr[] = { }; /* Cherryview-based platforms: CherryTrail and Braswell */ -static struct sst_acpi_mach sst_acpi_chv[] = { +static struct snd_soc_acpi_mach sst_acpi_chv[] = { { .id = "10EC5670", .drv_name = "cht-bsw-rt5672", diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 296b50640f46..c4d82ad41bd7 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -27,9 +27,9 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/da7213.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), @@ -224,7 +224,7 @@ static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ static int bytcht_da7213_probe(struct platform_device *pdev) { struct snd_soc_card *card; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int dai_index = 0; int ret_val = 0; @@ -243,7 +243,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6acacc0aeedb..8088396717e3 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -29,8 +29,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #include "../common/sst-dsp.h" struct byt_cht_es8316_private { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 3eeb02036aa1..f2c0fc415e52 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -32,9 +32,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5640.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #include "../common/sst-dsp.h" enum { @@ -737,7 +737,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { struct byt_rt5640_private *priv; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; int dai_index = 0; @@ -762,7 +762,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(byt_rt5640_codec_name, sizeof(byt_rt5640_codec_name), "%s%s", "i2c-", i2c_name); @@ -796,7 +796,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; struct acpi_buffer state = {0, NULL}; - struct sst_acpi_package_context pkg_ctx; + struct snd_soc_acpi_package_context pkg_ctx; bool pkg_found = false; state.length = sizeof(chan_package); @@ -808,7 +808,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) pkg_ctx.state = &state; pkg_ctx.data_valid = false; - pkg_found = sst_acpi_find_package_from_hid(mach->id, &pkg_ctx); + pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, + &pkg_ctx); if (pkg_found) { if (chan_package.aif_value == 1) { dev_info(&pdev->dev, "BIOS Routing: AIF1 connected\n"); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1dad5c98c9ef..51de1bbe8ecd 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -30,9 +30,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5651.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" enum { BYT_RT5651_DMIC_MAP, @@ -443,7 +443,7 @@ static char byt_rt5651_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { struct byt_rt5651_private *priv; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; int dai_index = 0; @@ -468,7 +468,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(byt_rt5651_codec_name, sizeof(byt_rt5651_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 83c923c39393..18d129caa974 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -32,9 +32,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5645.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI1 "rt5645-aif1" @@ -523,7 +523,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_cht_mc_probe(struct platform_device *pdev) { struct snd_soc_card *card = snd_soc_cards[0].soc_card; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct cht_mc_private *drv; const char *i2c_name = NULL; bool found = false; @@ -566,7 +566,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "%s%s", "i2c-", i2c_name); @@ -599,7 +599,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* format specified: 2 64-bit integers */ struct acpi_buffer format = {sizeof("NN"), "NN"}; struct acpi_buffer state = {0, NULL}; - struct sst_acpi_package_context pkg_ctx; + struct snd_soc_acpi_package_context pkg_ctx; bool pkg_found = false; state.length = sizeof(chan_package); @@ -611,7 +611,8 @@ static int snd_cht_mc_probe(struct platform_device *pdev) pkg_ctx.state = &state; pkg_ctx.data_valid = false; - pkg_found = sst_acpi_find_package_from_hid(mach->id, &pkg_ctx); + pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, + &pkg_ctx); if (pkg_found) { if (chan_package.aif_value == 1) { dev_info(&pdev->dev, "BIOS Routing: AIF1 connected\n"); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index d135640702e9..f8f21eee9b2d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -24,9 +24,10 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/soc-acpi.h> #include "../../codecs/rt5670.h" #include "../atom/sst-atom-controls.h" -#include "../common/sst-acpi.h" + /* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ #define CHT_PLAT_CLK_3_HZ 19200000 @@ -383,7 +384,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; struct cht_mc_private *drv; - struct sst_acpi_mach *mach = pdev->dev.platform_data; + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; const char *i2c_name; int i; @@ -395,7 +396,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* fixup codec name based on HID */ if (mach) { - i2c_name = sst_acpi_find_name_from_hid(mach->id); + i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); if (i2c_name) { snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", i2c_name); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 1a35149bcad7..2c958ec3f4fd 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,10 +1,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-match-objs := sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o -obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o obj-$(CONFIG_SND_SOC_INTEL_SST_FIRMWARE) += snd-soc-sst-firmware.o diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 1285cc597b6b..9866998de8be 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -21,7 +21,7 @@ #include <linux/platform_device.h> #include "sst-dsp.h" -#include "sst-acpi.h" +#include <sound/soc-acpi.h> #define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000 #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 @@ -30,7 +30,7 @@ /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; - struct sst_acpi_mach *machines; + struct snd_soc_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -49,7 +49,7 @@ struct sst_acpi_priv { struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -59,7 +59,7 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; - struct sst_acpi_mach *mach = sst_acpi->mach; + struct snd_soc_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { @@ -85,7 +85,7 @@ static int sst_acpi_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct sst_acpi_priv *sst_acpi; struct sst_pdata *sst_pdata; - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; struct sst_acpi_desc *desc; struct resource *mmio; int ret = 0; @@ -99,7 +99,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return -ENODEV; desc = (struct sst_acpi_desc *)id->driver_data; - mach = sst_acpi_find_machine(desc->machines); + mach = snd_soc_acpi_find_machine(desc->machines); if (mach == NULL) { dev_err(dev, "No matching ASoC machine driver found\n"); return -ENODEV; @@ -179,7 +179,7 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } -static struct sst_acpi_mach haswell_machines[] = { +static struct snd_soc_acpi_mach haswell_machines[] = { { "INT33CA", "haswell-audio", "intel/IntcSST1.bin", NULL, NULL, NULL }, {} }; @@ -197,7 +197,7 @@ static struct sst_acpi_desc sst_acpi_haswell_desc = { .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach broadwell_machines[] = { +static struct snd_soc_acpi_mach broadwell_machines[] = { { "INT343A", "broadwell-audio", "intel/IntcSST2.bin", NULL, NULL, NULL }, { "RT5677CE", "bdw-rt5677", "intel/IntcSST2.bin", NULL, NULL, NULL }, {} @@ -217,7 +217,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; #if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) -static struct sst_acpi_mach baytrail_machines[] = { +static struct snd_soc_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, {} diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h deleted file mode 100644 index afe9b87b8bd5..000000000000 --- a/sound/soc/intel/common/sst-acpi.h +++ /dev/null @@ -1,82 +0,0 @@ -/* - * Copyright (C) 2013-15, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - */ - -#include <linux/stddef.h> -#include <linux/acpi.h> - -struct sst_acpi_package_context { - char *name; /* package name */ - int length; /* number of elements */ - struct acpi_buffer *format; - struct acpi_buffer *state; - bool data_valid; -}; - -#if IS_ENABLED(CONFIG_ACPI) -/* translation fron HID to I2C name, needed for DAI codec_name */ -const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); -bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx); -#else -static inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) -{ - return NULL; -} -static inline bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx) -{ - return false; -} -#endif - -/* acpi match */ -struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); - -/* acpi check hid */ -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]); - -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - /* ACPI ID for the matching machine driver. Audio codec for instance */ - const u8 id[ACPI_ID_LEN]; - /* machine driver name */ - const char *drv_name; - /* firmware file name */ - const char *fw_filename; - - /* board name */ - const char *board; - struct sst_acpi_mach * (*machine_quirk)(void *arg); - const void *quirk_data; - void *pdata; -}; - -#define SST_ACPI_MAX_CODECS 3 - -/** - * struct sst_codecs: Structure to hold secondary codec information apart from - * the matched one, this data will be passed to the quirk function to match - * with the ACPI detected devices - * - * @num_codecs: number of secondary codecs used in the platform - * @codecs: holds the codec IDs - * - */ -struct sst_codecs { - int num_codecs; - u8 codecs[SST_ACPI_MAX_CODECS][ACPI_ID_LEN]; -}; - -/* check all codecs */ -struct sst_acpi_mach *sst_acpi_codec_list(void *arg); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 4abbd90e394c..31d8634e8aa1 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -28,7 +28,7 @@ #include <linux/firmware.h> #include <linux/delay.h> #include <sound/pcm.h> -#include "../common/sst-acpi.h" +#include <sound/soc-acpi.h> #include <sound/hda_register.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> @@ -439,10 +439,10 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) { struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct platform_device *pdev; - struct sst_acpi_mach *mach = driver_data; + struct snd_soc_acpi_mach *mach = driver_data; int ret; - mach = sst_acpi_find_machine(mach); + mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(bus->dev, "No matching machine driver found\n"); return -ENODEV; @@ -878,27 +878,27 @@ static void skl_remove(struct pci_dev *pci) dev_set_drvdata(&pci->dev, NULL); } -static struct sst_codecs skl_codecs = { +static struct snd_soc_acpi_codecs skl_codecs = { .num_codecs = 1, .codecs = {"10508825"} }; -static struct sst_codecs kbl_codecs = { +static struct snd_soc_acpi_codecs kbl_codecs = { .num_codecs = 1, .codecs = {"10508825"} }; -static struct sst_codecs bxt_codecs = { +static struct snd_soc_acpi_codecs bxt_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; -static struct sst_codecs kbl_poppy_codecs = { +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { .num_codecs = 1, .codecs = {"10EC5663"} }; -static struct sst_codecs kbl_5663_5514_codecs = { +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { .num_codecs = 2, .codecs = {"10EC5663", "10EC5514"} }; @@ -907,7 +907,7 @@ static struct skl_machine_pdata cnl_pdata = { .use_tplg_pcm = true, }; -static struct sst_acpi_mach sst_skl_devdata[] = { +static struct snd_soc_acpi_mach sst_skl_devdata[] = { { .id = "INT343A", .drv_name = "skl_alc286s_i2s", @@ -917,7 +917,7 @@ static struct sst_acpi_mach sst_skl_devdata[] = { .id = "INT343B", .drv_name = "skl_n88l25_s4567", .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, .pdata = &skl_dmic_data }, @@ -925,14 +925,14 @@ static struct sst_acpi_mach sst_skl_devdata[] = { .id = "MX98357A", .drv_name = "skl_n88l25_m98357a", .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, .pdata = &skl_dmic_data }, {} }; -static struct sst_acpi_mach sst_bxtp_devdata[] = { +static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { { .id = "INT343A", .drv_name = "bxt_alc298s_i2s", @@ -942,13 +942,13 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .id = "DLGS7219", .drv_name = "bxt_da7219_max98357a_i2s", .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, }, {} }; -static struct sst_acpi_mach sst_kbl_devdata[] = { +static struct snd_soc_acpi_mach sst_kbl_devdata[] = { { .id = "INT343A", .drv_name = "kbl_alc286s_i2s", @@ -958,7 +958,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "INT343B", .drv_name = "kbl_n88l25_s4567", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, .pdata = &skl_dmic_data }, @@ -966,7 +966,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98357A", .drv_name = "kbl_n88l25_m98357a", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, .pdata = &skl_dmic_data }, @@ -974,7 +974,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98927", .drv_name = "kbl_r5514_5663_max", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_5663_5514_codecs, .pdata = &skl_dmic_data }, @@ -982,7 +982,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { .id = "MX98927", .drv_name = "kbl_rt5663_m98927", .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = sst_acpi_codec_list, + .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_poppy_codecs, .pdata = &skl_dmic_data }, @@ -995,7 +995,7 @@ static struct sst_acpi_mach sst_kbl_devdata[] = { {} }; -static struct sst_acpi_mach sst_glk_devdata[] = { +static struct snd_soc_acpi_mach sst_glk_devdata[] = { { .id = "INT343A", .drv_name = "glk_alc298s_i2s", @@ -1004,7 +1004,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { {} }; -static const struct sst_acpi_mach sst_cnl_devdata[] = { +static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { { .id = "INT34C2", .drv_name = "cnl_rt274", diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/soc-acpi.c similarity index 63% rename from sound/soc/intel/common/sst-match-acpi.c rename to sound/soc/soc-acpi.c index 56d26f36a3cb..f21df28bc28e 100644 --- a/sound/soc/intel/common/sst-match-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,5 +1,5 @@ /* - * sst_match_apci.c - SST (LPE) match for ACPI enumeration. + * soc-apci.c - support for ACPI enumeration. * * Copyright (c) 2013-15, Intel Corporation. * @@ -14,9 +14,9 @@ * more details. */ -#include "sst-acpi.h" +#include <sound/soc-acpi.h> -static acpi_status sst_acpi_find_name(acpi_handle handle, u32 level, +static acpi_status snd_soc_acpi_find_name(acpi_handle handle, u32 level, void *context, void **ret) { struct acpi_device *adev; @@ -34,12 +34,12 @@ static acpi_status sst_acpi_find_name(acpi_handle handle, u32 level, return AE_OK; } -const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) { const char *name = NULL; acpi_status status; - status = acpi_get_devices(hid, sst_acpi_find_name, NULL, + status = acpi_get_devices(hid, snd_soc_acpi_find_name, NULL, (void **)&name); if (ACPI_FAILURE(status) || name[0] == '\0') @@ -47,9 +47,9 @@ const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) return name; } -EXPORT_SYMBOL_GPL(sst_acpi_find_name_from_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); -static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, +static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level, void *context, void **ret) { unsigned long long sta; @@ -63,26 +63,27 @@ static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, return AE_OK; } -bool sst_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) +bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) { acpi_status status; bool found = false; - status = acpi_get_devices(hid, sst_acpi_mach_match, &found, NULL); + status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL); if (ACPI_FAILURE(status)) return false; return found; } -EXPORT_SYMBOL_GPL(sst_acpi_check_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid); -struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) +struct snd_soc_acpi_mach * +snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { - struct sst_acpi_mach *mach; + struct snd_soc_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (sst_acpi_check_hid(mach->id) == true) { + if (snd_soc_acpi_check_hid(mach->id) == true) { if (mach->machine_quirk == NULL) return mach; @@ -92,14 +93,14 @@ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines) } return NULL; } -EXPORT_SYMBOL_GPL(sst_acpi_find_machine); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_machine); -static acpi_status sst_acpi_find_package(acpi_handle handle, u32 level, - void *context, void **ret) +static acpi_status snd_soc_acpi_find_package(acpi_handle handle, u32 level, + void *context, void **ret) { struct acpi_device *adev; acpi_status status = AE_OK; - struct sst_acpi_package_context *pkg_ctx = context; + struct snd_soc_acpi_package_context *pkg_ctx = context; pkg_ctx->data_valid = false; @@ -137,37 +138,38 @@ static acpi_status sst_acpi_find_package(acpi_handle handle, u32 level, return AE_OK; } -bool sst_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], - struct sst_acpi_package_context *ctx) +bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], + struct snd_soc_acpi_package_context *ctx) { acpi_status status; - status = acpi_get_devices(hid, sst_acpi_find_package, ctx, NULL); + status = acpi_get_devices(hid, snd_soc_acpi_find_package, ctx, NULL); if (ACPI_FAILURE(status) || !ctx->data_valid) return false; return true; } -EXPORT_SYMBOL_GPL(sst_acpi_find_package_from_hid); +EXPORT_SYMBOL_GPL(snd_soc_acpi_find_package_from_hid); -struct sst_acpi_mach *sst_acpi_codec_list(void *arg) +struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) { - struct sst_acpi_mach *mach = arg; - struct sst_codecs *codec_list = (struct sst_codecs *) mach->quirk_data; + struct snd_soc_acpi_mach *mach = arg; + struct snd_soc_acpi_codecs *codec_list = + (struct snd_soc_acpi_codecs *) mach->quirk_data; int i; if (mach->quirk_data == NULL) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (sst_acpi_check_hid(codec_list->codecs[i]) != true) + if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true) return NULL; } return mach; } -EXPORT_SYMBOL_GPL(sst_acpi_codec_list); +EXPORT_SYMBOL_GPL(snd_soc_acpi_codec_list); MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Common ACPI Match module"); +MODULE_DESCRIPTION("ALSA SoC ACPI module"); -- 2.14.1

1 0

[alsa-devel] [RFC PATCH 0/8] Prepare Sound Open Firmware (SOF) driver integration
by Pierre-Louis Bossart 21 Oct '17

21 Oct '17

The Sound Open Firmware work [1] is progressing quickly and drivers for the 1.0 release will soon be submitted upstream. Before we do so, there is a new to reshuffle the ACPI integration and Intel Kconfig/Makefiles. This RFC is a preview of what's coming, feedback would be greatly appreciated. We already have three incompatible drivers (BYT/HSW/BDW, Atom/BYT/CHT, SKL+) for historical reasons, blindly adding yet another one would be silly. We need to avoid having the same ACPI routines and machine tables duplicated in 3 locations (soc/intel/common, soc/atom/sst and the upcoming soc/sof). The ACPI routines should be part of sound/soc since there is no real hardware dependency (ACPI is not limited to Intel). The ACPI table definitions and quirks are moved to a soc/intel/common module so that when new HIDs or quirks are added "legacy" and the SOF drivers are updated at once. The Makefiles/Kconfig are also modified to select the SOC(s) first and the machine driver(s) second. The existing code only exposes machine drivers which precludes reuse of machine drivers when SOF platform drivers are selected. SOF drivers will be exposed in parallel to the 'Intel SST' ones (with an exclusive choice needed), but machine drivers will be shared. These patches apply on top of broonie/for-next + all the patches I submitted this week (no dependency on SOF). For people with allergic reactions to patches over email, the code is here: https://github.com/plbossart/sound/commits/topic/acpi-updates This patchset does not address the case where two drivers are selected for the same ACPI HID. I just don't know of any technical solution to that problem. [1] https://www.alsa-project.org/main/index.php/Firmware Pierre-Louis Bossart (8): ASoC: move ACPI common code out of Intel/sst tree ASoC: Intel: common: use c99 syntax for ACPI/machine tables ASoC: ACPI: add new fields for SOF support ASoC: Intel: move all ACPI match tables to common module ASoC: Intel: add SOF firmare/topology file information ASoC: Intel: move machine drivers to dedicated KConfig ASoC: Intel: reorder boards Kconfig by chronological order ASoC: Intel: clarify Kconfig dependencies include/sound/soc-acpi-intel-match.h | 32 ++ include/sound/soc-acpi.h | 107 ++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 + sound/soc/intel/Kconfig | 299 ++------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 319 ++-------------- sound/soc/intel/boards/Kconfig | 286 +++++++++++++++ sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +- sound/soc/intel/boards/bytcr_rt5651.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/soc-acpi-intel-match.c | 400 +++++++++++++++++++++ sound/soc/intel/common/sst-acpi.c | 36 +- sound/soc/intel/common/sst-acpi.h | 82 ----- sound/soc/intel/skylake/skl.c | 40 +-- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 +-- 20 files changed, 965 insertions(+), 752 deletions(-) create mode 100644 include/sound/soc-acpi-intel-match.h create mode 100644 include/sound/soc-acpi.h create mode 100644 sound/soc/intel/boards/Kconfig create mode 100644 sound/soc/intel/common/soc-acpi-intel-match.c delete mode 100644 sound/soc/intel/common/sst-acpi.h rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) -- 2.9.3

4 23

[alsa-devel] [PATCH 0/8] ASoC: Prepare Sound Open Firmware (SOF) driver integration
by Pierre-Louis Bossart 21 Oct '17

21 Oct '17

The Sound Open Firmware work [1] is progressing quickly and drivers for the 1.0 release will soon be submitted upstream. Before we do so, there is a new to reshuffle the ACPI integration and Intel Kconfig/Makefiles. We already have three incompatible drivers (BYT/HSW/BDW, Atom/BYT/CHT, SKL+) for historical reasons, blindly adding yet another one would be silly. We need to avoid having the same ACPI routines and machine tables duplicated in 3 locations (soc/intel/common, soc/atom/sst and the upcoming soc/sof). The ACPI routines should be part of sound/soc since there is no real hardware dependency (ACPI is not limited to Intel). The ACPI table definitions and quirks are moved to a soc/intel/common module so that when new HIDs or quirks are added "legacy" and the SOF drivers are updated at once. The Makefiles/Kconfig are also modified to select the SOC(s) first and the machine driver(s) second. The existing code only exposes machine drivers which precludes reuse of machine drivers when SOF platform drivers are selected. SOF drivers will be exposed in parallel to the 'Intel SST' ones (with an exclusive choice needed), but machine drivers will be shared. These patches apply on top of broonie/for-next + all the patches I submitted this week (no dependency on SOF). For people with allergic reactions to patches over email, the code is here: https://github.com/plbossart/sound/commits/topic/acpi-updates6 This patchset does not address the case where two drivers are selected for the same ACPI HID. I just don't know of any technical solution to that problem. changes since initial RFC: rebased to v4.14-rc2, used kernel style doc for machine descriptors, removed useless fields, split tables for each SOC corrected missing newlines Pierre-Louis Bossart (8): ASoC: move ACPI common code out of Intel/sst tree ASoC: Intel: common: use c99 syntax for ACPI/machine tables ASoC: ACPI: add new fields for SOF support ASoC: Intel: move all ACPI match tables to common module ASoC: Intel: add SOF firmare/topology file information ASoC: Intel: move machine drivers to dedicated KConfig ASoC: Intel: reorder boards Kconfig by chronological order ASoC: Intel: clarify Kconfig dependencies include/sound/soc-acpi-intel-match.h | 32 +++ .../common/sst-acpi.h => include/sound/soc-acpi.h | 75 +++-- sound/soc/Kconfig | 3 + sound/soc/Makefile | 6 + sound/soc/intel/Kconfig | 299 ++----------------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 319 ++------------------- sound/soc/intel/boards/Kconfig | 286 ++++++++++++++++++ sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 11 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 196 +++++++++++++ sound/soc/intel/common/soc-acpi-intel-cht-match.c | 194 +++++++++++++ .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 64 +++++ sound/soc/intel/common/sst-acpi.c | 36 +-- sound/soc/intel/skylake/skl.c | 40 +-- .../{intel/common/sst-match-acpi.c => soc-acpi.c} | 56 ++-- 21 files changed, 963 insertions(+), 692 deletions(-) create mode 100644 include/sound/soc-acpi-intel-match.h rename sound/soc/intel/common/sst-acpi.h => include/sound/soc-acpi.h (34%) create mode 100644 sound/soc/intel/boards/Kconfig create mode 100644 sound/soc/intel/common/soc-acpi-intel-byt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cht-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c rename sound/soc/{intel/common/sst-match-acpi.c => soc-acpi.c} (63%) -- 2.11.0

4 16

[alsa-devel] [PATCH 0/2] ASoC: audio-graph-scu-card: tidyup for Example
by Kuninori Morimoto 21 Oct '17

21 Oct '17

Hi Mark These are not a big deal, but tidyup patches for audio-graph-scu-card Example. Kuninori Morimoto (2): ASoC: audio-graph-scu-card: add missing Capture routing on Example ASoC: audio-graph-scu-card: remove unnecessary route patch from Example 1 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: Intel: sst: remove redundant variable dma_dev_name
by Colin King 21 Oct '17

21 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The pointer dma_dev_name is assigned but never read, it is redundant and can therefore be removed. Cleans up clang warning: sound/soc/intel/common/sst-firmware.c:288:3: warning: Value stored to 'dma_dev_name' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/common/sst-firmware.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index a086c35f91bb..79a9fdf94d38 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -274,7 +274,6 @@ int sst_dma_new(struct sst_dsp *sst) struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; struct resource mem; - const char *dma_dev_name; int ret = 0; if (sst->pdata->resindex_dma_base == -1) @@ -285,7 +284,6 @@ int sst_dma_new(struct sst_dsp *sst) * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { case SST_DMA_TYPE_DW: - dma_dev_name = "dw_dmac"; break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", -- 2.14.1

3 2

[alsa-devel] Driver-alsa for Icon Producer Works-192
by Андрей Король 20 Oct '17

20 Oct '17

Hello ! Is it possible to support the sound card Icon Producer Works-192 in the alsa? More about the map: VIA ICEnsemble ICE1724HT (Envy24HT/PT) http://soundcard.hw-specs.com/ICON_Producer_192/ https://nson.org/komplektuyushhie/zvukovye-karty/icon-producer-works-192x.h…

1 0

Re: [alsa-devel] Add support for ALC236
by Takashi Iwai 20 Oct '17

20 Oct '17

On Fri, 20 Oct 2017 09:13:44 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch was support ALC236. Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH] ASoC: rt5616: fix 0x91 default value
by Bard Liao 20 Oct '17

20 Oct '17

The default value of register 0x91 is 0x0c00 instead of 0x0000. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index c94e94f..0e5f54a 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -98,7 +98,7 @@ static const struct reg_default rt5616_reg[] = { { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0000 }, - { 0x91, 0x0000 }, + { 0x91, 0x0c00 }, { 0x92, 0x0000 }, { 0x93, 0x2000 }, { 0x94, 0x0200 }, -- 2.7.4

1 0

[alsa-devel] [PATCH] tinycompress: clarify dual-license wording
by Pierre-Louis Bossart 20 Oct '17

20 Oct '17

There were reports of confusion in legal circles introduced by the wording "tinycompress is provided under LGPL and BSD dual license" in the README. Clarify with the addition of the wording traditionally used for dual licenses: "This file is provided under a dual BSD/LGPLv2.1 license. When using or redistributing this file, you may do so under either license" Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- COPYING | 3 +++ README | 3 ++- include/tinycompress/tinycompress.h | 3 +++ include/tinycompress/tinymp3.h | 3 +++ include/tinycompress/version.h | 3 +++ src/lib/compress.c | 3 +++ src/utils/cplay.c | 3 +++ src/utils/crecord.c | 3 +++ 8 files changed, 23 insertions(+), 1 deletion(-) diff --git a/COPYING b/COPYING index f963926..dfc7426 100644 --- a/COPYING +++ b/COPYING @@ -1,3 +1,6 @@ +tinycompress is provided under a dual BSD/LGPLv2.1 license. When using or + redistributing tinycompress, you may do so under either license. + BSD LICENSE tinycompress library for compress audio offload in alsa diff --git a/README b/README index 73c19a6..689ca75 100644 --- a/README +++ b/README @@ -22,7 +22,8 @@ Http: http://git.alsa-project.org/?p=tinycompress.git alsa-devel(a)alsa-project.org. 4. LICENSE - tinycompress is provided under LGPL and BSD dual license + tinycompress is provided under a dual BSD/LGPLv2.1 license. When using or + redistributing tinycompress, you may do so under either license. 5. CREDITS - Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> for library design diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..780dce6 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * Copyright (c) 2011-2012, Intel Corporation diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..b0d429d 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * mp3 header and prasing diff --git a/include/tinycompress/version.h b/include/tinycompress/version.h index 000f6a2..82a3e6c 100644 --- a/include/tinycompress/version.h +++ b/include/tinycompress/version.h @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * Copyright (c) 2011-2012, Intel Corporation diff --git a/src/lib/compress.c b/src/lib/compress.c index 84738d2..8ae6bbd 100644 --- a/src/lib/compress.c +++ b/src/lib/compress.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * tinycompress library for compress audio offload in alsa diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 032a971..e0603fd 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * tinyplay command line player for compress audio offload in alsa diff --git a/src/utils/crecord.c b/src/utils/crecord.c index 1fd3698..e051c2d 100644 --- a/src/utils/crecord.c +++ b/src/utils/crecord.c @@ -1,4 +1,7 @@ /* + * This file is provided under a dual BSD/LGPLv2.1 license. When using or + * redistributing this file, you may do so under either license. + * * BSD LICENSE * * crecord command line recorder for compress audio record in alsa -- 2.11.0

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[alsa-devel] [PATCH 0/2] rt5651: Enable jack detection
by Carlo Caione 20 Oct '17

20 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> Small set of patches to enable jack detection for the rt5651 codec. Please note that: - The jack detection is restricted to the JD1_1 pin only (the only hardware I have uses this pin for jack detection) - AFAICT (reading the rt5651 datasheet) there is no way to detect runtime the type of the jack inserted as done for the others rt56* codecs, so in the driver I always assume that we are dealing with an headset. Carlo Caione (2): ASoC: rt5651: Enable jack detection on JD1_1 ASoC: intel: byt: Add headset jack include/sound/rt5651.h | 7 +++ sound/soc/codecs/rt5651.c | 91 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5651.h | 3 ++ sound/soc/intel/boards/bytcr_rt5651.c | 24 +++++++++ 4 files changed, 123 insertions(+), 2 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH] ALSA: hda: Avoid racy recreation of widget kobjects
by Takashi Iwai 19 Oct '17

19 Oct '17

The refresh of HD-audio widget sysfs kobjects via snd_hdac_refresh_widget_sysfs() is slightly racy. The driver recreates the whole tree from scratch after deleting the whole. When CONFIG_DEBUG_KOBJECT_RELEASE option is used, kobject release doesn't happen immediately but delayed, while the re-creation of the same named kobject happens soon after invoking kobject_put(). This may end up with the conflicts of duplicated kobjects, as found in the bug report below. In this patch, we take another approach to refresh the tree: instead of recreating the whole tree, just add the new nodes and delete the non-existing nodes. Since the refresh happens only once at initialization, no longer race would happen. Along with the code change, merge snd_hdac_refresh_widget_sysfs() with the existing snd_hdac_refresh_widgets() with an additional bool flag for simplifying the code. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=197307 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 3 +-- sound/hda/hdac_device.c | 43 ++++++++++--------------------------------- sound/hda/hdac_sysfs.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/hda/local.h | 2 ++ sound/pci/hda/hda_codec.c | 2 +- 5 files changed, 61 insertions(+), 36 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 96546b30e900..6f1118545bb8 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -111,8 +111,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size); -int snd_hdac_refresh_widgets(struct hdac_device *codec); -int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); +int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs); unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..06f845e293cb 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -87,7 +87,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, fg = codec->afg ? codec->afg : codec->mfg; - err = snd_hdac_refresh_widgets(codec); + err = snd_hdac_refresh_widgets(codec, false); if (err < 0) goto error; @@ -388,11 +388,12 @@ static void setup_fg_nodes(struct hdac_device *codec) /** * snd_hdac_refresh_widgets - Reset the widget start/end nodes * @codec: the codec object + * @sysfs: re-initialize sysfs tree, too */ -int snd_hdac_refresh_widgets(struct hdac_device *codec) +int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) { hda_nid_t start_nid; - int nums; + int nums, err; nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid); if (!start_nid || nums <= 0 || nums >= 0xff) { @@ -401,6 +402,12 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec) return -EINVAL; } + if (sysfs) { + err = hda_widget_sysfs_reinit(codec, start_nid, nums); + if (err < 0) + return err; + } + codec->num_nodes = nums; codec->start_nid = start_nid; codec->end_nid = start_nid + nums; @@ -408,36 +415,6 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets); -/** - * snd_hdac_refresh_widget_sysfs - Reset the codec widgets and reinit the - * codec sysfs - * @codec: the codec object - * - * first we need to remove sysfs, then refresh widgets and lastly - * recreate it - */ -int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec) -{ - int ret; - - if (device_is_registered(&codec->dev)) - hda_widget_sysfs_exit(codec); - ret = snd_hdac_refresh_widgets(codec); - if (ret) { - dev_err(&codec->dev, "failed to refresh widget: %d\n", ret); - return ret; - } - if (device_is_registered(&codec->dev)) { - ret = hda_widget_sysfs_init(codec); - if (ret) { - dev_err(&codec->dev, "failed to init sysfs: %d\n", ret); - return ret; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_hdac_refresh_widget_sysfs); - /* return CONNLIST_LEN parameter of the given widget */ static unsigned int get_num_conns(struct hdac_device *codec, hda_nid_t nid) { diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 42d61bf41969..e123ba6b2e81 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -414,3 +414,50 @@ void hda_widget_sysfs_exit(struct hdac_device *codec) { widget_tree_free(codec); } + +int hda_widget_sysfs_reinit(struct hdac_device *codec, + hda_nid_t start_nid, int num_nodes) +{ + struct hdac_widget_tree *tree; + hda_nid_t end_nid = start_nid + num_nodes; + hda_nid_t nid; + int i; + + if (!codec->widgets) + return hda_widget_sysfs_init(codec); + + tree = kmemdup(codec->widgets, sizeof(*tree), GFP_KERNEL); + if (!tree) + return -ENOMEM; + + tree->nodes = kcalloc(num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); + if (!tree->nodes) { + kfree(tree); + return -ENOMEM; + } + + /* prune non-existing nodes */ + for (i = 0, nid = codec->start_nid; i < codec->num_nodes; i++, nid++) { + if (nid < start_nid || nid >= end_nid) + free_widget_node(codec->widgets->nodes[i], + &widget_node_group); + } + + /* add new nodes */ + for (i = 0, nid = start_nid; i < num_nodes; i++, nid++) { + if (nid < codec->start_nid || nid >= codec->end_nid) + add_widget_node(tree->root, nid, &widget_node_group, + &tree->nodes[i]); + else + tree->nodes[i] = + codec->widgets->nodes[nid - codec->start_nid]; + } + + /* replace with the new tree */ + kfree(codec->widgets->nodes); + kfree(codec->widgets); + codec->widgets = tree; + + kobject_uevent(tree->root, KOBJ_CHANGE); + return 0; +} diff --git a/sound/hda/local.h b/sound/hda/local.h index 0d5bb159d538..c586ea62d49e 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -28,6 +28,8 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) extern const struct attribute_group *hdac_dev_attr_groups[]; int hda_widget_sysfs_init(struct hdac_device *codec); +int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid, + int num_nodes); void hda_widget_sysfs_exit(struct hdac_device *codec); #endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3db26c451837..bb086665c3e0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -977,7 +977,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; - err = snd_hdac_refresh_widget_sysfs(&codec->core); + err = snd_hdac_refresh_widgets(&codec->core, true); if (err < 0) return err; -- 2.14.2

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[alsa-devel] [PATCH v2] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 19 Oct '17

19 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Due to bugs in BIOS it's possible that the HDA capability link list is not constructed properly. This may lead to driver going into unknown state. So whenever driver discovers unknown HDA capability, log it as error and stop traversing the link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

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[alsa-devel] [BUG] aplay doesn't honor `-t raw` when WAVE header is detected on input
by Nutchanon Wetchasit 19 Oct '17

19 Oct '17

Hello, I ran into a problem on how alsa-utils's aplay command handles `-t` option when I was trying to play a PCM WAVE file in loop, using following command: ( while true; do cat audio.wav; done ) | aplay -t raw -f S16_LE -r 44100 -c 2 An "audio.wav" can be any 44100 Hz 16-bit stereo PCM WAVE file, and `-t raw` was specially specified to make aplay reject length information in the header. However, once the command is run, first line of the output would read: Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Which shows that aplay still read the input as WAVE format, despite the `-t raw` option being explicitly specified. And you would notice that the audio *does not loop*. Instead, it will play just once, followed by silence. However, if WAVE header was stripped out (or corrupted), aplay would now treat the input as raw audio stream, and play a continuously repeated sound as expected: ( while true; do dd if=audio.wav of=/dev/stdout ibs=44 skip=1; done ) | \ aplay -t raw -f S16_LE -r 44100 -c 2 The first line of output would also show the stream as "raw data" too: Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo This problem exists on the latest alsa-utils 1.1.4, as well as the older 1.0.25 which came with my GNU/Linux distribution; though I'm not certain whether this is also applicable to other file format (like Sun .AU) or not. Please investigate. Regards, Nutchanon Wetchasit alsa-utils: 1.1.4 (source distribution) libasound: 1.1.4 (source distribution) System: Debian GNU/Linux 7.0 "Wheezy" i386 P.S. The "audio.wav" I used in my testing is a siren sound generated by Sox: sox -b 16 -c 2 -e signed-integer -r 44100 -n audio.wav \ synth 0.5 sine 800 gain -6 : synth 0.5 sine 600 gain -6

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[alsa-devel] [PATCH v2 0/2] ALSA: hda - Fix incorrect TLV callback check
by Takashi Iwai 18 Oct '17

18 Oct '17

Hi, this is a revised patchset for addressing the incorrect TLV callback check introduced by a set_fs() removal commit. A fix for a superfluous '-' in the code is split as another fix so that it can be applied to older kernels without the rest. Other than that, only the changelog text updates since the older patch (that was embedded in the mail thread "4.13 regression: get_kctl_0dB_offset doesn't handle all possible callbacks"). thanks, Takashi === Takashi Iwai (2): ALSA: hda: Remove superfluous '-' added by printk conversion ALSA: hda - Fix incorrect TLV callback check introduced during set_fs() removal include/sound/control.h | 3 ++ sound/core/vmaster.c | 31 +++++++++++++++ sound/pci/hda/hda_codec.c | 97 +++++++++++++++++++++++++++-------------------- 3 files changed, 89 insertions(+), 42 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 18 Oct '17

18 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> When an invalid capability is discovered, stop traversing the capability link list further. print the invalid capability error and set it to zero Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..8f7d0d9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; break; } -- 2.7.4

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