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

[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

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[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

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[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

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[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

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[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

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