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

[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…

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[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

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[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

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[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] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 24 Feb '20

24 Feb '20

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..80e4757176d8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

2 1

[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] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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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] alsa-lib-1.1.6.tar.bz2: some files are missing in the tarball
by Lars Wendler 13 May '19

13 May '19

Hello list, please have a look at this Gentoo bug report [1] where a user reported that he cannot compile alsa-lib-1.1.6 with --enable-alisp configure option. The error message is: make[3]: *** No rule to make target 'SI7018/sndoc-mixer.alisp', needed by 'all-am'. Stop. This happens because the directory src/conf/cards/SI7018 is missing as well as the following files: src/conf/cards/aliases.alisp src/conf/smixer.conf src/conf/sndo-mixer.alisp First I thought those files were removed intentionally but checking the alsa-lib git tree I can see all files still being available. So at Gentoo we fixed this by re-adding the files through a patch [2]. Any chance this was an accidental oversight? Kind regards Lars [1] https://bugs.gentoo.org/652422 [2] https://gitweb.gentoo.org/repo/gentoo.git/tree/media-libs/alsa-lib/files/al… P.S.: Please CC me on replies as I am not subscribed to the list. -- Lars Wendler Gentoo package maintainer GPG: 21CC CF02 4586 0A07 ED93 9F68 498F E765 960E 9B39

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[alsa-devel] [PATCH v2] ASoC: core: Allow topology to override machine driver FE DAI link config.
by Liam Girdwood 03 May '19

03 May '19

Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1. o Now iterate over component_list to fix crash with DT enumerated cards. o Make sure name prefix is only added once with deferred probiing. include/sound/soc.h | 13 ++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ 3 files changed, 123 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..b1d65fcb8756 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -806,6 +806,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -963,6 +971,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1002,6 +1013,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1011,6 +1023,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..38bf7d01894b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -852,6 +852,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1461,7 +1464,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1512,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1543,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1870,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1947,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 153218aa4909..db51849ae9bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -865,8 +865,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.17.1

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[alsa-devel] [PATCH] ASoC: rockchip: add missing INTERLEAVED PCM attribute
by Katsuhiro Suzuki 24 Apr '19

24 Apr '19

This patch adds SNDRV_PCM_INFO_INTERLEAVED into PCM hardware info. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index f77538319221..9e7b5fa4cf59 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 1, -- 2.18.0

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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] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

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[alsa-devel] snd_hda_codec_hdmi: `hdaudio hdaudioC0D2: Unable to bind the codec`
by Paul Menzel 18 Feb '19

18 Feb '19

Dear Linux folks, With drm-tip (v4.19-rc3-946-g09b295662edd) Linux prints `snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec` to the log. ``` […] [ 12.481788] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC671: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 12.482536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) [ 12.483184] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 12.483737] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.484198] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.484582] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 12.485100] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 12.485519] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 12.497685] snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec [ 12.498311] hdaudio hdaudioC0D2: Unable to bind the codec [ 12.498789] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input6 [ 12.499567] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input7 [ 12.500268] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input8 [ 12.501039] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 12.501683] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 17.743338] [drm:intel_pch_type [i915]] Found SunrisePoint PCH […] ``` Is that an error or just a notice? Kind regards, Paul

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

3 4

[alsa-devel] [PATCH 0/6] ASoC: rt5651: Various fixes and quirks
by Hans de Goede 24 Dec '18

24 Dec '18

Hi All, Here is my last (big) round of (Intel) rt5651 support updates, with these patches combined with the updated, matching UCM profiles from here: https://github.com/jwrdegoede/alsa-lib/commits/master All 7 different devices I have work 100% OOTB, without requiring any manual config or mixer setup. After this there should be no more big changes. I will submit the matching alsa-lib UCM profile changes upstream once this set is in -next (and thus I can be sure the long-names will not change due to review). Note that unfortunately my previous round of cleanup patches removed 1 input mapping (of the 3 mappings in total it removed) which is actually necessary. So this series re-introduces that 1 mapping. I did not notice this before because UCM input profile switching does not work reliably for some reason, something which I still need to investigate. This time I manually veryfied that the correct / expected mixer settings where used while testing. Regards, Hans

4 23

[alsa-devel] [PATCH v2 01/13] conf/ucm: bytcr-rt5645: Use the generic bytcr/PlatformEnableSeq.conf
by Hans de Goede 28 Nov '18

28 Nov '18

Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is identical to all the cset statements this commit removes. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/chtrt5645/HiFi.conf | 116 +------------------------------ 1 file changed, 1 insertion(+), 115 deletions(-) diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf index e81866cf..d993f6ae 100644 --- a/src/conf/ucm/chtrt5645/HiFi.conf +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -11,121 +11,7 @@ SectionVerb { EnableSequence [ cdev "hw:chtrt5645" - # 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 (used if ssp2 is connected to aif1) - 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" - - # modem_out settings (used if ssp0 is connected to aif2) - cset "name='modem_out mix 0 codec_in0 Switch' off" - cset "name='modem_out mix 0 codec_in1 Switch' off" - cset "name='modem_out mix 0 media_loop1_in Switch' off" - cset "name='modem_out mix 0 media_loop2_in Switch' off" - cset "name='modem_out mix 0 pcm0_in Switch' on" - cset "name='modem_out mix 0 pcm1_in Switch' off" - cset "name='modem_out mix 0 sprot_loop_in Switch' off" - cset "name='modem_out Gain 0 Switch' on" - cset "name='modem_out Gain 0 Volume' 0" - - # input settings - # pcm1_out settings - - # input used when SSP2 is connected - cset "name='codec_in0 Gain 0 Switch' on" - cset "name='codec_in0 Gain 0 Volume' 0" - - # input used when SSP0 is connected - cset "name='modem_in Gain 0 Switch' on" - cset "name='modem_in Gain 0 Volume' 0" - - cset "name='pcm1_out mix 0 codec_in0 Switch' on" - cset "name='pcm1_out mix 0 modem_in 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" + <bytcr/PlatformEnableSeq.conf> # Output Configuration cset "name='DAC1 L Mux' IF1 DAC" -- 2.18.0

6 30

Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 27 Nov '18

27 Nov '18

Please don't drop Cc to ML. Also avoid top-posting. On Sat, 04 Aug 2018 19:02:53 +0200, David Ulricht wrote: > > How can I do that ? > I tried verifying if /proc/asound/card0/codec#0 changes when i put > handphone jack in and out with "diff ver1 ver2" and it is exactly the same. Just do for each bit as you tried for GPIO output. In this case, the direction bit should be different (set 0), and you need to read. > > One strange thing I note is hda-jack-retask.fw contains: > [pincfg] > 0x24 0x90100080 > 0x25 0x90100082 > I have 0x90100080 for 0x24 but in /proc/asound/card0/codec#0 : > Node 0x24 [Pin Complex] wcaps 0x400101: Stereo > Pincap 0x00000010: OUT > Pin Default 0x90100110: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Connection: 1 > 0x02 > > Shouldn't "Pin Default 0x90100110" be 0x90100080 after applying the patch ? No, the codec config value itself won't change, the driver uses the given value only internally. Takashi > I can see that Applying patch firmware 'hda-jack-retask.fw' is before > hdaudioC0D0: autoconfig for Generic: line_outs=2 (0x24/0x25/0x0/0x0/0x0) > type:speaker > in dmesg and this worries me, maybe the pins are overriden by generic > cirrus logic ? > > > On Sat, Aug 4, 2018 at 7:26 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 17:44:59 +0200, > > David Ulricht wrote: > > > > > > I used the following script running as root to verify the 8 GPIOs while > > > playing long youtube video, > > > I have running pulseaudio meanwhile and not restarting it, i think its > > okay > > > like that people on the ubuntu forums say that after using > > > hda-verb and setting a gpio active immediately sound appears, so i guess > > > there is no need to restart pulseaudio. > > > I have done the same test with headphones in. > > > No sound in both scenarios, no input no output detected (i'm watching > > > pavucontrol for input detection). > > > > This codec is fairly unique and doesn't provide the standard jack > > detection on each pin. Maybe it's via either GPIO or any other > > method. You can try to read GPIO pins (set as input) to see whether > > any of them corresponds to the jack detection, for example. > > > > > > Takashi > > > > > > > > #!/usr/bin/env python3 > > > import os,time > > > for x in range(0,256): > > > mhex=("0x%0.2X" % x) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > > > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > > > time.sleep(3) > > > > > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > > > IO[0-7] changes its configuration, > > > in comparison Joel Krahemann in his scripts is changing the hex right > > after > > > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > > > correctly. > > > > > > I found the following post: > > > https://forum.manjaro.org/t/sound-not-working-on-a-2017- > > imac-27-5k-retina/43638 > > > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 > > has > > > a similar soundcard. > > > How can i setup with command line or any other utility: > > > """select “off” in the top menu that propose to setup the Digital > > > Surround""" > > > this is some setting in xfce4-mixer package which is not available for > > > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > > > find this setting in audio mixer of xfce4 either. > > > > > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > > > David Ulricht wrote: > > > > > > > > > > Hello, > > > > > > > > > > Macbook pro models containing CS8409 for sure are: > > > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > > > > > I'm attaching working sound configuration from Windows 10 registry > > (note > > > > > that the cs420x might need to be ignored or might be important i > > don't > > > > > really know). I believe what is really used is in the > > PinConfigOverride > > > > > section. > > > > > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > > > and your override (user_pin_cfg), there aren't so many changes. > > > > All changes are pretty minor, and I guess it won't influence on the > > > > driver behavior. > > > > > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > > > Or it's some direct I2C control. The INI file mentions it. > > > > And that's above HD-audio driver's responsibility. > > > > > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > > > > > It's because the codec chip has no output amplifier at all. > > > > So there can be neither output volume nor mute control on this chip. > > > > > > > > > What I have tried is execute: > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > > > putting @@ from 0x00 to 0x50 > > > > > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > > > 0x02, 0x04, 0x08, ... 0x80. > > > > > > > > And how is the current situation? You can't play *and* record > > > > anything from any inputs / outputs? > > > > > > > > > > > > Takashi > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [PATCH 1/2] ASoC: max9892x: Add documentation for reset-gpio support
by Cheng-Yi Chiang 20 Nov '18

20 Nov '18

max98927 codec driver will support reset-gpio binding so it can toggle reset line in its probe function. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/max9892x.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt index f6171591ddc61..8bf30483eedcd 100644 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ b/Documentation/devicetree/bindings/sound/max9892x.txt @@ -29,6 +29,8 @@ Required properties: Range : 0 (off), 1 (on), Default : 0 - reg : the I2C address of the device for I2C + - reset-gpio: reset gpio the chip is connected to. This is supported + on max98927. Example: @@ -38,4 +40,5 @@ codec: max98927@3a { imon-slot-no = <1>; interleave-mode = <0>; reg = <0x3a>; + reset-gpio = <&tlmm 75 GPIO_ACTIVE_LOW>; }; -- 2.16.4

6 13

Re: [alsa-devel] Controlling the Tascam FW-1884
by Takashi Sakamoto 16 Nov '18

16 Nov '18

Hi Scott, (C.Ced to alsa-devel, ffado-devel) On Sep 7 2018 23:01, Scott Bahling wrote: > I recently got my hands on a FW-1884 and would like to get it working in > Linux. I'm mostly interested in using the control interface which should > support several midi control interface protocols like Mackie Control or > Mackie HUI. From the user manual it appears that the host computer needs to > set the registers on the FW-1884 to switch modes (they reference Windows > applications). > > I see that you have recently been working on user space support asynchronous > commands to such devices. The hinawa_utils code has some beginnings of > support for setting options on the Tascam devices and I have just started > playing with that. Thanks for your suggestion. At present, the most of features on control surface of FW-1884 are not available, especially for fader, knob and start/stop buttons. In data transmission between system and devices on IEEE 1394 bus, both of asynchronous transactions and isochronous packet streams are available. In a case of TASCAM FireWire series, messages via MIDI physical interface is bypassed to asynchronous transaction, while messages generated on control surface are bypassed to isochronous packets, multiplexed with PCM frames. When enabling multiplexing messages to isochronous packet, demultiplexer should be implemented to ALSA firewire-tascam driver, but not yet. You can see my initial RFC in alsa-devel[1]. This patch enables userspace applications to perform the demultiplexing, however this idea easily brings cache-coherency problems due to page mapping between kernel/userspace and impossible to be applied. We need to investigate another idea to achieve the demultiplexing in kernel land. As another option, according to page 26 in 'TASCAM FW-1884 Owner's Manual', when programming Keys to port 1-4, system could receive MIDI messages from control surface of FW-1884. However, I've not investigated it. > How are you determining the registers and flags? Do you have documentation, > or are you reverse engineering? Do you know the registers and flags for > setting the control protocol? > > If you have any hints on how figure out the proper registers and flags I'm > happy to contribute to the work you have started and help support more > functions of the FW-1884. User manuals are documents to which we can refer. The most vendors licenses their drivers/software with EULA and in most cases it can't allow users to binary-based reverse engineering. All of my work are done with external protocol analyzer unit on IEEE 1394 bus. When working to implement drivers, I read/classified/parsed content of many packets sniffed by the analyzer. An initial idea of libhinawa/hinawa-utils is to assist this my work. I heard FFADO developers uses some ways for this type of work: - Usage of host controller of TI PCILynx and driver module (nosy.ko) - Windows application; bushound[2] However, this type of work takes you long time, so I don't necessarily recommend you. It's 2018 and I think you can find alternative control surfaces on USB. If you take long time to make devices available on GNU/Linux, it's better to select devices on 'live' buses instead of 'legacy' buses such as IEEE 1394 bus, IMO. In a case of USB, wireshark with usbcap is the easiest way for us to sniff packets on Windows. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://perisoft.net/bushound/ Regards Takashi Sakamoto

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Re: [alsa-devel] [linux-sunxi] [BUG] On Allwinner A20 Board Cubietruck distorted SPDIF opical output
by Markus Mitsch 28 Oct '18

28 Oct '18

hi. i posted my answer wrong so here is my finding: i made the output work by disabling the sigma delta modulation. after compiling the kernel i can output through spdif with any rate. greetings markus 2018-05-26 18:38 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: > 2018-05-26 16:57 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >> 2018-05-23 17:12 GMT+02:00 Code Kipper <codekipper(a)gmail.com>: >>> Hi All, >>> I can see that the problem was introduced somewhere here >>> >>> DOESN'T WORK >>> ffc3eb6f3a83 Merge tag 'sunxi-clk-for-4.15' of >>> https://git.kernel.org/pub/scm/linux/kernel/git/sunxi/linux into >>> clk-next >>> ae74ac082886 Merge tag 'clk-v4.15-exynos-pm' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> faa865f18cb7 Merge tag 'clk-v4.15-samsung' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> f09a6b86fdae Merge tag 'clk-renesas-for-v4.15-tag1' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/geert/renesas-drivers >>> into clk-next >>> 4328a2186e51 clk: sunxi-ng: sun4i: Export video PLLs >>> 553c7d5ba2fe clk: sunxi-ng: Add A83T display clocks >>> 7679eb20353d clk: samsung: Add a separate driver for Exynos4412 ISP clocks >>> 8ca8ac102484 clk: samsung: Add dt bindings for Exynos4412 ISP clock controller >>> 75920aac275a clk: samsung: Instantiate Exynos4412 ISP clocks only when available >>> 24ea78a09f89 clk: sunxi-ng: sun8i: a23: Use sigma-delta modulation for audio PLL >>> ee6501d69217 clk: sunxi-ng: sun6i: Use sigma-delta modulation for audio PLL >>> 042f7f8f9715 clk: sunxi-ng: sun5i: Use sigma-delta modulation for audio PLL >>> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >>> a5e3e2b2ef85 clk: sunxi-ng: sun8i: h3: Use sigma-delta modulation for audio PLL >>> 392ba5fafcdf clk: sunxi-ng: nm: Add support for sigma-delta modulation >>> 05d2eaac96d4 clk: sunxi-ng: Add sigma-delta modulation support >>> 4cdbc40d64d4 clk: sunxi-ng: nm: Check if requested rate is supported >>> by fractional clock >>> d51fe3ba9773 clk: sunxi-ng: sun5i: Fix bit offset of audio PLL post-divider >>> 4e8975cbb516 clk: samsung: exynos5433: mark PM functions as __maybe_unused >>> a4f21e9ceb5c clk: samsung: Remove obsolete clkdev alias support >>> 45d882daf88a clk: samsung: Add explicit MPLL, EPLL clkdev aliases in >>> S3C2443 driver >>> cd05417f728b clk: samsung: Rework clkdev alias handling in S3C2443 driver >>> efea8d377184 clk: samsung: Rework clkdev alias handling in Exynos5440 driver >>> 36ba48240b19 clk: samsung: Drop useless alias in Exynos5420 clk driver >>> 29964890f31c clk: samsung: Remove clkdev alias support in Exynos5250 clk driver >>> 6de08891c896 clk: samsung: Remove double assignment of CLK_ARM_CLK in >>> Exynos4 driver >>> 58f4a5ff3a62 clk: samsung: Remove clkdev alias support in Exynos4 clk driver >>> c9194fb623b0 clk: samsung: Remove support for obsolete Exynos4212 CPU clock >>> d5cd103b06f9 clk: samsung: Remove support for Exynos4212 SoCs in >>> Exynos CLKOUT driver >>> 7c4f63ec94a1 clk: sunxi-ng: a83t: Fix invalid csi-mclk mux offset >>> WORKS >>> 0a4e632b6f9d Merge branch 'clk-fixes' into clk-next >>> >>> BR, >>> CK >>> >>> >>> On 22 May 2018 at 21:27, Code Kipper <codekipper(a)gmail.com> wrote: >>>> Hi Markus, >>>> I've been able to reproduce this on my A31 board with linux-next(plays >>>> at 44100 but not at 96K)so it is an issue. I also tested the same code >>>> on my A64 and it worked fine, The blocks are very similar so it maybe >>>> a clocking issue, I'll look into this. >>>> BR, >>>> CK >>>> >>>> On 21 May 2018 at 20:17, Markus Mitsch <markusmitsch86(a)googlemail.com> wrote: >>>>> 2018-05-21 15:17 GMT+02:00 Maxime Ripard <maxime.ripard(a)bootlin.com>: >>>>>> On Mon, May 21, 2018 at 02:43:55PM +0200, Markus Mitsch wrote: >>>>>>> 2018-05-21 11:43 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >>>>>>> >>> Hello, >>>>>>> >>> >>>>>>> >>> i have a new problem. if i want to output sound from optical i get >>>>>>> >>> distorted sound. At 44.1 khz its like an old vinly, but at 48 khz and >>>>>>> >>> higher it is very bad. thanks in advance for your time. >>>>>>> > >>>>>>> >> Hi Markus, >>>>>>> >> I haven't tested it for a while but when I did it worked as high as >>>>>>> >> 96K with no issues. Does it work ok with the original cubietruck >>>>>>> >> software?, >>>>>>> >> CK >>>>>>> > >>>>>>> > Hello, >>>>>>> > i am using arch linux with mainline kernel 4.16.8. parhaps i can try >>>>>>> > today with official image. >>>>>>> > mfg >>>>>>> >>>>>>> Hi. >>>>>>> So, with debian image from 2014 with 3.4.79 kernel the sound is clear >>>>>>> up to 96 khz. >>>>>> >>>>>> You said that it is a new problem, when was it introduced? Could you >>>>>> bisect the issue? >>>>>> >>>>>> Maxime >>>>> >>>>> Hello. >>>>> I dont know since when this problem appears. When saying "new" i >>>>> refered to my older post. I tried to look into the dts file and the >>>>> code >>>>> but i am afraid this is a bit too high for me. I can say that the >>>>> analog output works, even at high sampling rates. >>>>> And I have read something about a clock source could be the problem. >>>>> I'm sorry I can't tell you any more. >>>>> >>>>> mfg >>>>> Markus >> >> Hello, >> i tried undoing part of the commit >> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >> by setting the value for the divider from 1 to 4 again on both places. >> with no success. >> >> markus > > i now had a little success. the output works!! > i disabled the sigma-delta support by making > ccu_sdm_helper_enable in drivers/clk/sunxi-ng/ccu_sdm.c return imediately. > > markus

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[alsa-devel] [PATCH v4 00/14] ASoC: Add support to WCD9335 Audio Codec
by srinivas.kandagatla＠linaro.org 24 Oct '18

24 Oct '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou for reviewing v3 patchset, here is v4 addressing comments from v3. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v3 (https://lkml.org/lkml/2018/9/4/318/) - Updated mbhc dt bindings according to Robs Comments. - Added a cleanup patch to common up fixups. - Addressed various comments on mfd driver by Lee Jones. - Added acks on bindings and irq support. - fixed module loading Srinivas Kandagatla (14): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support ASoC: qcom: common: move be_hw_fixup to common .../devicetree/bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 13 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 394 ++ include/linux/mfd/wcd9335/registers.h | 640 +++ include/linux/mfd/wcd9335/wcd9335.h | 81 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 +++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5241 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 128 +- sound/soc/qcom/common.c | 17 + sound/soc/qcom/sdm845.c | 22 +- 14 files changed, 7168 insertions(+), 35 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.9.3

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[alsa-devel] [asoc:for-4.20 333/424] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 23 Oct '18

23 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 1c8bc7b3de5e76cb89aacdc7be1475a028af505f commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/424] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by David Lin 19 Oct '18

19 Oct '18

Add driver for NAU88C22. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8822.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8822.c | 1136 +++++++++++++++++ sound/soc/codecs/nau8822.h | 204 +++ 5 files changed, 1363 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000000..a471d162d4e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +NAU8822 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8822" + + - reg : the I2C address of the device. + +Example: + +codec: nau8822@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..57365dbc1581 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -1299,6 +1300,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..181a7d222fd7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -366,6 +367,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..622ce947f134 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1136 @@ +/* + * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver + * + * Copyright 2017 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * Based on WM8974.c + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Playback Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Playback Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Playback Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 1), + + SOC_DOUBLE_R_TLV("PGA Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX2_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..aa79c969cd44 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,204 @@ +/* + * nau8822.h -- NAU8822 Soc Audio Codec driver + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * 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. + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ -- 2.18.0

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Re: [alsa-devel] [PATCH 1/3] MIPS: jz4780: Allow access to jz4740-i2s
by Paul Burton 17 Oct '18

17 Oct '18

Hi Matthieu, On Wed, Jun 06, 2018 at 09:38:08PM +0200, Mathieu Malaterre wrote: > diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig > index 1a354a6b6e87..35d82d96e781 100644 > --- a/sound/soc/jz4740/Kconfig > +++ b/sound/soc/jz4740/Kconfig > @@ -1,20 +1,20 @@ > config SND_JZ4740_SOC > - tristate "SoC Audio for Ingenic JZ4740 SoC" > - depends on MACH_JZ4740 || COMPILE_TEST > + tristate "SoC Audio for Ingenic JZ4740/JZ4780 SoC" > + depends on MACH_JZ4740 || MACH_JZ4780 || COMPILE_TEST Perhaps this could be MACH_INGENIC, or even just MIPS? Thanks, Paul

2 4

[alsa-devel] [ANNOUNCE] 2018 Linux Audio miniconference
by Mark Brown 16 Oct '18

16 Oct '18

Hi, As in previous years we're trying to organize an audio miniconference so we can get together and talk through issues, especially design decisons, face to face. This year's event will be held on Sunday October 21st in Edinburgh, the day before ELC Europe starts there. We're still finalizing the venue (we expect to be able to confirm next week) but it'll be outside the ELC venue so there will be no need to purchase an ELC ticket to attend. 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! If you're planning to attend please fill out the form here: https://docs.google.com/spreadsheets/d/11p2pSNxS7sJysg1JwErWi0T2Dq7XKRtAGng… Thanks, Mark

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[alsa-devel] Fixing EAPD for STAC9202 on an embedded platform.
by Andy Davison 11 Oct '18

11 Oct '18

Hi everyone, I’m hoping that somebody here might be able to spot where I’m going wrong; I'm working with a picture-frame-like device manufactured by the late OpenPeak Inc. known as the OpenFrame (or O2 Joggler in the UK). It has an internal speaker and a 3.5mm line out socket; none of the other connections to its STAC9202 codec chip are exposed to the end user. I've been porting Ubuntu Bionic to the device and everything is done, except for the audio driver. Back in kernel 3.18 it was possible to use the following kernel patch and userland ALSA patch to have these two outputs work correctly, with the insertion of a plug into the line out properly muting the internal speaker. https://github.com/andydvsn/OpenFrame-Kernel/blob/master/patches/03-stac920… https://github.com/andydvsn/OpenFrame-Ubuntu/blob/master/overlay-trusty/lib… In kernel 4.14 the same trick no longer works, so I've rewritten the patch to more properly add support for this variant of STAC9202 to ALSA. It now looks like this: https://gist.github.com/andydvsn/7bffa8de0ed691b14508485d75501ffc This seems to line up the pin configurations appropriately, but when a plug is inserted to the line out, although the audio to the internal speaker is muted, the speaker emits a crackling sound whenever the device is processing. It seems that it is not being properly powered down and is amplifying general circuit noise. Disconnecting the plug returns sound to the internal speaker and the crackling ceases. This is reliable across multiple devices. Here is the alsa-info output from both the crackling kernel 4.14 and the working-just-fine kernel 3.18. https://gist.github.com/andydvsn/ccf55e6f7961dd405fc39d41a6ce9ef6 I don't have sufficient knowledge of ALSA drivers and patches and have been working purely from the configurations used by others over the years, so I would really appreciate somebody looking at the alsa-info output and the new patch to determine where I should go from here. I believe this is all to do with EAPD control, but what options I have for controlling that aren't known to me right now. Ideally, I want to end up with the audio muting and switching correctly when the jack is inserted. A bonus would be to lose the unused (or at least inaccessible) audio input and output options in alsamixer and gain individual control of the internal speaker and line out levels (which appear to be paired together at the moment). All the best, A. -- Andrew Davison

2 11

[alsa-devel] [PATCH 00/11] Add ZxR support + bugfixes
by Connor McAdams 08 Oct '18

08 Oct '18

This patch series adds support for the Sound Blaster ZxR, as well as a few bug fixes. This should be the last ca0132 based Creative card that needed support to be added. Also, I did check to make sure each patch compiles properly this time, but you can check yourself just to be sure. :) Connor McAdams (11): ALSA: hda/ca0132 - Fix AE-5 control type ALSA: hda/ca0132 - Fix surround sound with output effects ALSA: hda/ca0132 - Add ZxR quirks + new quirk check function ALSA: hda/ca0132 - Add ZxR pincfg ALSA: hda/ca0132 - Add DBpro hda_codec_ops ALSA: hda/ca0132 - Add ZxR init commands ALSA: hda/ca0132 - Add ZxR DSP post-download commands ALSA: hda/ca0132 - Add ZxR input/output select commands ALSA: hda/ca0132 - Remove input select enum for ZxR ALSA: hda/ca0132 - Add ZxR 600 ohm gain control ALSA: hda/ca0132 - Add ZxR exit commands sound/pci/hda/patch_ca0132.c | 369 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 345 insertions(+), 24 deletions(-) -- 2.7.4

3 17

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Connor McAdams 08 Oct '18

08 Oct '18

I might not get around to writing the fix myself, but, here's the COEF's from within Windows. I've marked the ones missing inside the main init function. So, if anyone else wants to add these in, feel free. On Wed, Sep 12, 2018 at 1:16 PM, Håvard <hovardslill(a)gmail.com> wrote: > That was without running the program (sorry) > > This is output when running program: > > Before there was no "Current verb...." > They came when I opened up windows microphone settings. (something I didn't > do before) > > -Håvard > > Den ons. 12. sep. 2018 kl. 19:11 skrev Håvard <hovardslill(a)gmail.com>: >> >> I think this is right! >> >> I first booted up, then tested speakers (That I could not hear) and then >> tested microphone (Which worked perfectly) in windows settings. >> >> (My first email was dissaproved as it was too large) >> >> Den ons. 12. sep. 2018 kl. 19:05 skrev Håvard <hovardslill(a)gmail.com>: >>> >>> I think this is right! >>> >>> I first booted up, then tested speakers (That I could not hear) and then >>> tested microphone (Which worked perfectly) in windows settings. >>> >>> -Håvard >>> >>> Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams >>> <conmanx360(a)gmail.com>: >>>> >>>> Also, sorry to keep sending more replies, make sure the only trace you >>>> have on is vfio_region_write, and not the read one. The read spams up >>>> the console. >>>> >>>> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >>>> wrote: >>>> > Reason being, it may not have identified the CORB buffer properly or >>>> > something. That's very odd to have no data. Never seen that before. >>>> > >>>> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams >>>> > <conmanx360(a)gmail.com> wrote: >>>> >> Could you try again and make a copy of your terminal output? >>>> >> >>>> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> >>>> >> wrote: >>>> >>> Hi again. >>>> >>> Sorry for not noticing all files were empty, did it not run for long >>>> >>> enough? >>>> >>> >>>> >>> -Håvard >>>> >>> >>>> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard >>>> >>> <hovardslill(a)gmail.com>: >>>> >>>> >>>> >>>> Sorry my bad again it was working perfectly, just missed a step. >>>> >>>> >>>> >>>> I'll add the dumps as an attatchment. I had both external headset >>>> >>>> and >>>> >>>> microphone plugged in and let the VM run in the background for >>>> >>>> ~30min while >>>> >>>> doing other stuff. I did not play music or test my microphone in >>>> >>>> the VM >>>> >>>> either, If i did anything wrong, please tell me! >>>> >>>> >>>> >>>> It stopped at 0x104f4 >>>> >>>> >>>> >>>> -Håvard >>>> >>>> >>>> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >>>> >>>> <conmanx360(a)gmail.com>: >>>> >>>>> >>>> >>>>> Hm.... that's odd. They should show up in the folder you ran the >>>> >>>>> command from. Does your console show any "DumpMem entered..." or >>>> >>>>> something like that? You may have a permissions error. I've had >>>> >>>>> some >>>> >>>>> people report that as an issue before. >>>> >>>>> >>>> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >>>> >>>>> wrote: >>>> >>>>> > Hi! >>>> >>>>> > >>>> >>>>> > Took a while, but I think I got it working. Did not see any >>>> >>>>> > "frame[xx]" >>>> >>>>> > files though. What dumps do you want? >>>> >>>>> > >>>> >>>>> > -Håvard >>>> >>>>> > >>>> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard >>>> >>>>> > <hovardslill(a)gmail.com>: >>>> >>>>> >> >>>> >>>>> >> Sorry about that, I forgot to enable a kernel config... >>>> >>>>> >> >>>> >>>>> >> -Håvard >>>> >>>>> >> >>>> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >>>> >>>>> >> <conmanx360(a)gmail.com>: >>>> >>>>> >>> >>>> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >>>> >>>>> >>> controls. >>>> >>>>> >>> You can check what module is loaded by doing lspci -v . It >>>> >>>>> >>> should say >>>> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the >>>> >>>>> >>> vfio-bind >>>> >>>>> >>> script >>>> >>>>> >>> before trying it? >>>> >>>>> >>> >>>> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard >>>> >>>>> >>> <hovardslill(a)gmail.com> >>>> >>>>> >>> wrote: >>>> >>>>> >>> > Thank you for the offer! >>>> >>>>> >>> > >>>> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >>>> >>>>> >>> > seem to >>>> >>>>> >>> > bind my >>>> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >>>> >>>>> >>> > to >>>> >>>>> >>> > function as >>>> >>>>> >>> > normal when I (think I) have bound it to a stud. >>>> >>>>> >>> > >>>> >>>>> >>> > -Håvard >>>> >>>>> >>> > >>>> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >>>> >>>>> >>> > <conmanx360(a)gmail.com>: >>>> >>>>> >>> >> >>>> >>>>> >>> >> One thing you could try, is using the program I used to >>>> >>>>> >>> >> reverse >>>> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. >>>> >>>>> >>> >> If your >>>> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >>>> >>>>> >>> >> you >>>> >>>>> >>> >> could >>>> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >>>> >>>>> >>> >> capture >>>> >>>>> >>> >> the commands. I'd be willing to look through the dumps to >>>> >>>>> >>> >> see if >>>> >>>>> >>> >> there >>>> >>>>> >>> >> are any special verbs or anything. >>>> >>>>> >>> >> >>>> >>>>> >>> >> You can find the program here: >>>> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >>>> >>>>> >>> >> >>>> >>>>> >>> >> Let me know. >>>> >>>>> >>> >> >>>> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard >>>> >>>>> >>> >> <hovardslill(a)gmail.com> >>>> >>>>> >>> >> wrote: >>>> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response >>>> >>>>> >>> >> > before I >>>> >>>>> >>> >> > read >>>> >>>>> >>> >> > your >>>> >>>>> >>> >> > last one. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >>>> >>>>> >>> >> > headset >>>> >>>>> >>> >> > multijacks. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the >>>> >>>>> >>> >> > tip! >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > -Håvard >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >>>> >>>>> >>> >> > <hovardslill(a)gmail.com>: >>>> >>>>> >>> >> > >>>> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Do you know where I can ask around for more help on the >>>> >>>>> >>> >> >> issue? >>>> >>>>> >>> >> >> I >>>> >>>>> >>> >> >> don't >>>> >>>>> >>> >> >> want to give up yet. >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> -Håvard >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >>>> >>>>> >>> >> >> <tiwai(a)suse.de>: >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >>>> >>>>> >>> >> >>> Håvard wrote: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >>>> >>>>> >>> >> >>> > automatically >>>> >>>>> >>> >> >>> > changes >>>> >>>>> >>> >> >>> > to >>>> >>>>> >>> >> >>> it >>>> >>>>> >>> >> >>> > in pavucontrol. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Then it likely requires some additional initialization >>>> >>>>> >>> >> >>> outside >>>> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >>>> >>>>> >>> >> >>> vendor-specific. >>>> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >>>> >>>>> >>> >> >>> Windows >>>> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Takashi >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >>>> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >>>> >>>>> >>> >> >>> > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Thank you for replying! >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic >>>> >>>>> >>> >> >>> > > > Playback >>>> >>>>> >>> >> >>> > > > Volume" >>>> >>>>> >>> >> >>> > > > in >>>> >>>>> >>> >> >>> > > hdaanalyzer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >>>> >>>>> >>> >> >>> > > > made white noise come from the headset. However >>>> >>>>> >>> >> >>> > > > it did >>>> >>>>> >>> >> >>> > > > not >>>> >>>>> >>> >> >>> > > > change or >>>> >>>>> >>> >> >>> > > react >>>> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >>>> >>>>> >>> >> >>> > > > physically. >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >>>> >>>>> >>> >> >>> > > > either >>>> >>>>> >>> >> >>> > > > alsamixer >>>> >>>>> >>> >> >>> or >>>> >>>>> >>> >> >>> > > > hdaanalyzer. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > It's a mixer mute switch. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > The microphone works perfectly fine under >>>> >>>>> >>> >> >>> > > > Windows, so I >>>> >>>>> >>> >> >>> > > > don't >>>> >>>>> >>> >> >>> > > > think >>>> >>>>> >>> >> >>> it is >>>> >>>>> >>> >> >>> > > > the mic pin. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >>>> >>>>> >>> >> >>> > > wrong >>>> >>>>> >>> >> >>> > > pin, >>>> >>>>> >>> >> >>> > > too. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Takashi >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > -Håvard >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi >>>> >>>>> >>> >> >>> > > > Iwai >>>> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >>>> >>>>> >>> >> >>> >: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >>>> >>>>> >>> >> >>> > > > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Additional relevant info: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three >>>> >>>>> >>> >> >>> > > > > > years ago >>>> >>>>> >>> >> >>> > > > > > on >>>> >>>>> >>> >> >>> > > > > > Sun >>>> >>>>> >>> >> >>> > > > > > Jun >>>> >>>>> >>> >> >>> > > 17:15:54 >>>> >>>>> >>> >> >>> > > > > CEST >>>> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >>>> >>>>> >>> >> >>> > > > > > but did >>>> >>>>> >>> >> >>> > > > > > not >>>> >>>>> >>> >> >>> > > > > > touch >>>> >>>>> >>> >> >>> on the >>>> >>>>> >>> >> >>> > > > > > external microphone problem: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > alsa-info.sh: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >>>> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help >>>> >>>>> >>> >> >>> > > > > > the >>>> >>>>> >>> >> >>> > > > > > issue. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > There are several of similar bug reports >>>> >>>>> >>> >> >>> > > > > > around the >>>> >>>>> >>> >> >>> > > > > > web >>>> >>>>> >>> >> >>> experiencing >>>> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >>>> >>>>> >>> >> >>> > > > > > seems to >>>> >>>>> >>> >> >>> > > > > > happen >>>> >>>>> >>> >> >>> > > > > > with >>>> >>>>> >>> >> >>> all >>>> >>>>> >>> >> >>> > > > > laptops >>>> >>>>> >>> >> >>> > > > > > under the G751 name. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >>>> >>>>> >>> >> >>> > > > > and >>>> >>>>> >>> >> >>> > > > > unmute/adjust >>>> >>>>> >>> >> >>> "Mic >>>> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >>>> >>>>> >>> >> >>> > > > > you >>>> >>>>> >>> >> >>> > > > > hear >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from >>>> >>>>> >>> >> >>> > > > > NID 0x18 >>>> >>>>> >>> >> >>> > > > > to >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> mixer >>>> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >>>> >>>>> >>> >> >>> > > > > outputs. >>>> >>>>> >>> >> >>> > > > > So >>>> >>>>> >>> >> >>> > > > > this >>>> >>>>> >>> >> >>> > > > > can >>>> >>>>> >>> >> >>> be >>>> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from >>>> >>>>> >>> >> >>> the ext >>>> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >>>> >>>>> >>> >> >>> > > > > something >>>> >>>>> >>> >> >>> > > > > outside >>>> >>>>> >>> >> >>> > > > > HD-audio codec. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > Takashi >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> > _______________________________________________ >>>> >>>>> >>> >> > Alsa-devel mailing list >>>> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 26

[alsa-devel] [PATCH v2 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 05 Oct '18

05 Oct '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v1 (https://patchwork.kernel.org/cover/10585805/) - added support to snd_compr_set_runtime_buffer() - make use of generic copy function - few trivial cleanups suggested by Vinod Koul Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 377 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 443 insertions(+), 4 deletions(-) -- 2.19.0

2 9

[alsa-devel] Problem and Patch for B&W PX Bluetooth Headphones
by Nicolas Herochao 04 Oct '18

04 Oct '18

Hello, I'm writing this mail because of a Crash when connecting the B&W PX headphones via usb. Half a year ago this used to work with Linux after setting the default sample rate of pulseaudio to 48000k. A Launchpad Bug seems to describe this exact issue (detailed description below): https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1773697. It pointed me to the right direction, and today I have successfully compiled and tested a small patch that fixes the errors while also setting the sample rate to 48000k, making these headphones plug&play on any newer firmware. They sound awesome via USB! I would like to submit this patch, but it would be my first contribution and it could potentially break the B&W PX on older firmware. I can't test it since I see no obvious ways to downgrade the firmware. The automatic detection of these headphones always worked except for the sample-rate and perhaps the worst result is that old firmware will show a non-functional microphone, like on Windows. So I decided to first present the patch on this mailing list and ask for feedback and help. If anyone wants to pick the patch up, I'd appreciate that as well. My questions: * Can a quirk somehow handle multiple firmware versions of a usb device? * If not: Will this quirk allow old firmware to work out of the box but show a non-functional microphone? (And is that okay?) Thanks for your time and any feedback you can provide! Sincerely, Nicolas PS: more detail on the cause of the issue: ------ A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active with the following dmesg errors: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) --------------- Finally, here is the patch (based on the current mainline kernel and generated according to all Guidelines I could find) which fixes the quirk by removing the microphone part: ---------------- From 748587283db85e219902abf4ac012864005943c8 Mon Sep 17 00:00:00 2001 From: Nicolas Huaman <nicolas(a)herochao.de> Date: Fri, 28 Sep 2018 18:05:15 +0200 Subject: [PATCH 1/1] SOUND: update quirk for B&W PX to remove microphone Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

2 1

[alsa-devel] [PATCH 0/7] Add devres support to card resources
by Takashi Iwai 04 Oct '18

04 Oct '18

Hi, here is the patchset containing the changes to support a device-managed resource management for ALSA card and memory pages. The actual code changes are identical with the previous RFC series, but a few documentation updates are included in this official set. Takashi === Takashi Iwai (7): ALSA: core: Add device-managed page allocator helper ALSA: core: Add managed card creation ALSA: intel8x0: Allocate resources with device-managed APIs ALSA: atiixp: Allocate resources with device-managed APIs ALSA: hda: Allocate resources with device-managed APIs ALSA: doc: Brush up the old writing-an-alsa-driver ALSA: doc: Add device-managed resource section .../kernel-api/writing-an-alsa-driver.rst | 340 ++++++++++-------- include/sound/core.h | 5 + include/sound/memalloc.h | 4 + sound/core/init.c | 95 ++++- sound/core/memalloc.c | 88 ++++- sound/pci/atiixp.c | 104 ++---- sound/pci/atiixp_modem.c | 104 ++---- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 43 +-- sound/pci/intel8x0.c | 143 +++----- sound/pci/intel8x0m.c | 143 +++----- 11 files changed, 514 insertions(+), 556 deletions(-) -- 2.18.0

2 19

[alsa-devel] Thread spinning in kernel snd_pcm_link()/snd_pcm_unlink()
by Rob Duncan 02 Oct '18

02 Oct '18

I'm trying to address a bug where we end up with a thread spinning and consuming an entire cpu. The issue seems to be this code in sound/core/pcm_native.c: /* Writer in rwsem may block readers even during its waiting in queue, * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to * spin until it gets the lock. */ static inline void down_write_nonblock(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) cond_resched(); } The original commit for this is 67ec1072b053c15564e6090ab30127895dc77a89 What we're suspecting is that a normal thread (SCHED_OTHER) has a reader lock and a real-time thread using SCHED_RR or SCHED_FIFO is trying to take the writer lock. If both threads are pinned to the same CPU for some reason then the reader thread will never get scheduled (because the real-time writer thread is still runnable), and we will never make progress. Does this sound right? What can we do to fix this? Thanks, Rob.

2 2

[alsa-devel] [PATCH alsa-lib] ioplug: Fix comment in struct snd_pcm_ioplug
by Andre Guedes 02 Oct '18

02 Oct '18

This patch fixes the comment about 'pcm' field in 'struct snd_pcm_ioplug' which wrongly refers to snd_pcm_extplug_create(). Signed-off-by: Andre Guedes <andre.guedes(a)intel.com> --- include/pcm_ioplug.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm_ioplug.h b/include/pcm_ioplug.h index b16fc8b8..81ac8613 100644 --- a/include/pcm_ioplug.h +++ b/include/pcm_ioplug.h @@ -105,7 +105,7 @@ struct snd_pcm_ioplug { */ void *private_data; /** - * PCM handle filled by #snd_pcm_extplug_create() + * PCM handle filled by #snd_pcm_ioplug_create() */ snd_pcm_t *pcm; -- 2.14.4

3 2

[alsa-devel] [PATCH - Multiple plugins 0/4] Fix leaks on multiple plugins
by Andre Guedes 02 Oct '18

02 Oct '18

Hi all, While I was working on AAF plugin RFC v2 series [1], I spotted several leaks in some plugins when handling *_hw_constraint() error cases. This series fixes these issues. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… Andre Guedes (4): a52: Fix leaks when a52_set_hw_constraint() fails jack: Fix leaks when jack_set_hw_constraint() fails oss: Fix leaks when oss_hw_constraint() fails usb_stream: Fix leaks when us_set_hw_constraint() fails a52/pcm_a52.c | 2 +- jack/pcm_jack.c | 1 + oss/pcm_oss.c | 2 +- usb_stream/pcm_usb_stream.c | 1 + 4 files changed, 4 insertions(+), 2 deletions(-) -- 2.14.4

3 6

[alsa-devel] [PATCH] ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL
by Takashi Sakamoto 02 Oct '18

02 Oct '18

In blocking mode of IEC 61883-1/6, when one isochronous packet includes data for events, the data is for the same number of events as SYT_INTERVAL decided according to sampling transmission frequency (SFC). IEC 61883-1/6 engine of ALSA firewire stack applies constraints of period and buffer size of PCM intermediate buffer of PCM substream. At present, this constraint is designed to round the size up/down to 32 frames. This value comes from the least common multiple (LCM) of SYT_INTERVAL. Although this looks to work well, in lower sampling rate, applications are not allowed to set size of period quite near period time constraint (at present 5 msec per period). This commit adds PCM rules for period/buffer size and rate to obsoletes the constraints based on LCM. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 78 +++++++++++++++++++++++++++++++---- 1 file changed, 71 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index cb9acfe60f6a..fcd965f1d69e 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = { }; EXPORT_SYMBOL(amdtp_rate_table); +static int apply_constraint_to_size(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *s = hw_param_interval(params, rule->var); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = { + .min = s->min, .max = s->max, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int rate = amdtp_rate_table[i]; + unsigned int step = amdtp_syt_intervals[i]; + + if (!snd_interval_test(r, rate)) + continue; + + t.min = roundup(t.min, step); + t.max = rounddown(t.max, step); + } + + if (snd_interval_checkempty(&t)) + return -EINVAL; + + return snd_interval_refine(s, &t); +} + +static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int step = amdtp_syt_intervals[i]; + unsigned int rate = amdtp_rate_table[i]; + + if (s->min % step || s->max % step) + continue; + + t.min = min(t.min, rate); + t.max = max(t.max, rate); + } + + return snd_interval_refine(r, &t); +} + /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. - * - * TODO: These constraints can be improved with proper rules. - * Currently apply LCM of SYT_INTERVALs. */ - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) goto end; - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); end: return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH][alsa-lib] timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 6041cab2..4d35c84d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -750,7 +750,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ed0a120d4f08..404d4b9ffe76 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -752,7 +752,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix mismatch for register mask and value in ext controller.
by Keyon Jie 02 Oct '18

02 Oct '18

E.g. for snd_hdac_ext_bus_link_power_up(), we should set mask to be AZX_MLCTL_SPA(it was 0), and AZX_MLCTL_SPA as value to power up it, here correct it and several similar mismatches. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 5bc4a1d587d4..60cb00fd0c69 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); @@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, AZX_PPCTL_PIE); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); @@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) */ int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, - AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; -- 2.17.1

2 1

[alsa-devel] Re-reading alsa-lib configuration when opening sound device
by Pavel Hofman 02 Oct '18

02 Oct '18

Hi, I am playing with virtual balanced inputs https://www.diyaudio.com/forums/pc-based/326747-virtual-balanced-regular-so… . For proper functionality it needs a regular recalibration of the ttable multiplication coefficients of route plugin in .asoundrc. The audio stack is alsa -> wine -> Arta measurement software. When I stop measuring in arta (without exiting the software), the soundcard gets closed - confirmed in /proc/asound/.../hw_params. Now I change the ttable coeffs. After re-opening the device (continuing with measurement in the running arta instance) the changes in .asoundrc are ignored. I have to restart the arta application for the changes to take effect. When I run e.g. aplay inbetween, it accepts the changes right away. Upon continuing in arta, the original ttable coeffs are still loaded for this instance. I understand it is the way wine handles the device somewhere in https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.c . Please when is the alsa-lib config loaded by the library/application? Is there any way to force a config reload from outside? Would e.g. calling snd_config_update before snd_pcm_open on line https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.… help? If so, what overhead does it impose? Perhaps I could ask the wine guys to consider such change if it worked and required minimum overhead. I very much appreciate any help. Best regards, Pavel.

2 4

[alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones
by Thomas Gläßle 02 Oct '18

02 Oct '18

Hi, on both my desktop and laptop (archlinux), `alsactl monitor` beginns spinning at 100% cpu when I unplug my USB headset. This happens every time and can be reproduced as follows: - plug in USB headphones - start `alsactl monitor` - unplug headphones It happens only if the headphones are already plugged in when starting alsactl. The increased activity does not lead to spamming stdout or similar, alsactl continues to perform seemingly correct, i.e. reports when I change volume levels etc, but nothing apart from the ordinary. I'm sorry if this is not the right place to report this issue or if the problem is already, I couldn't find anything about it. Below is the output from alsa-info.sh: with headphones: http://www.alsa-project.org/db/?f=0dfc632e3adafb8bc78a182e0a225bcb13bea9fa after unplugging: http://www.alsa-project.org/db/?f=41dbf0735fdd7cd2416a6b4a602505715d25b44f Please let me know if you need further info. Best regards, Thomas

2 2

[alsa-devel] [PATCH linux-next 10/10] ASoC: rsnd: Add kctrl to configure dai's busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> ssi0, ssi1, ssi2, ssi3, ssi4 and ssi9 have busif0 to busif7, currently only busif0 is used by all ssi. But tdm split and tdm ex-split mode, which require busif1 to busif7. This patch adds kctrl in ssi for dai-link, to make it possible to configure the busif used by each dai-link. Also adds rsnd_dma_addr_update() interface to make it possible to update dma data address when there is change in condition of dai connection. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/dma.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 31 ++++++++++++++++++++++++++++++- 3 files changed, 64 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..8c284fd73fc1 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,38 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, + struct rsnd_mod *dma_mod) +{ + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(dma_mod); + + if (!dmac) + return; + + rsnd_dma_of_path(mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 59b6d89d8edc..6c16d8403f7a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -233,6 +233,8 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, struct rsnd_mod *dma_mod); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4ac4b5b75ae2..b469db238697 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -160,6 +160,15 @@ int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) return io->busif->val; } +static void rsnd_ssi_set_busif(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + if (!rsnd_ssi_use_busif(io)) + return; + + rsnd_dma_addr_update(io, mod, io->dma); +} + static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -718,6 +727,10 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) { + int is_play = rsnd_io_is_play(io); + int id = rsnd_mod_id(mod); + int ret = 0; + /* * rsnd_rdai_is_clk_master() will be enabled after set_fmt, * and, pcm_new will be called after it. @@ -725,7 +738,23 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, */ rsnd_ssi_parent_attach(mod, io); - return 0; + if (!rsnd_ssi_use_busif(io)) + return 0; + + if (rsnd_ssi_is_multi_slave(mod, io)) + return 0; + + /* SSI5 to SSI8 only have one BUSIF */ + if (id <= 4 || id == 9) + ret = rsnd_kctrl_new_s(mod, io, rtd, + is_play ? + "SSI Out BUSIF" : + "SSI In BUSIF", + rsnd_kctrl_accept_anytime, + rsnd_ssi_set_busif, + io->busif, 7); + + return ret; } static int rsnd_ssi_common_probe(struct rsnd_mod *mod, -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 09/10] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> SSI may use different busif for data transfer, this patch adds busif property to each dai stream, to indicate the busif used by playback/capture stream. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 16 +++++++++++++++- sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 5 ++++- 3 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..aee3132c72b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -560,6 +560,14 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +struct rsnd_kctrl_cfg_s *rsnd_busif_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= 2 * rsnd_rdai_nr(priv))) + return NULL; + + return priv->busif + id; +} + static struct snd_soc_dai_driver *rsnd_daidrv_get(struct rsnd_priv *priv, int id) { @@ -1045,6 +1053,9 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, io_playback = &rdai->playback; io_capture = &rdai->capture; + io_playback->busif = rsnd_busif_get(priv, dai_i); + io_capture->busif = rsnd_busif_get(priv, rsnd_rdai_nr(priv) + dai_i); + snprintf(rdai->name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", dai_i); rdai->priv = priv; @@ -1108,6 +1119,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct snd_soc_dai_driver *rdrv; struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; int nr; int is_graph; int dai_i; @@ -1123,12 +1135,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) rdrv = devm_kcalloc(dev, nr, sizeof(*rdrv), GFP_KERNEL); rdai = devm_kcalloc(dev, nr, sizeof(*rdai), GFP_KERNEL); - if (!rdrv || !rdai) + busif = devm_kcalloc(dev, 2 * nr, sizeof(*busif), GFP_KERNEL); + if (!rdrv || !rdai || !busif) return -ENOMEM; priv->rdai_nr = nr; priv->daidrv = rdrv; priv->rdai = rdai; + priv->busif = busif; /* * parse all dai diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..59b6d89d8edc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -456,6 +456,7 @@ struct rsnd_dai_stream { struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ + struct rsnd_kctrl_cfg_s *busif; u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) @@ -629,6 +630,8 @@ struct rsnd_priv { */ struct snd_soc_dai_driver *daidrv; struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; + int rdai_nr; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 75a3d15084c2..4ac4b5b75ae2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,10 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif->val; } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 00/10] ASoC: rsnd: support to set different busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Rcar audio DMA/SSIU already support to use different BUSIF to transfer data, but busif is fixed at 0. This patch set adds kctrl interface "SSI Out/In BUSIF", which enables to configure busif for each dai-stream used in transfer. device tree files are updated to use extended audio dmac register, and add dma request number for busif1 ~ busif7 Jiada Wang (10): ASoC: rsnd: ssi: Request dedicated dma channels for busif1 to 7 arm64: r8a7795: add dma request number for busif1 ~ busif7 arm64: r8a7796: add dma request number for busif1 ~ busif7 arm64: r8a774a1: add dma request number for busif1 ~ busif7 arm64: r8a7795: change to use extended audio dmac register arm64: r8a7796: change to use extended audio dmac register arm64: r8a774a1: change to use extended audio dmac register arm64: r8a77965: change to use extended audio dmac register ASoC: rsnd: add busif property to dai stream ASoC: rsnd: Add kctrl to configure dai's busif arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 139 ++++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 132 +++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 138 +++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- sound/soc/sh/rcar/core.c | 16 ++- sound/soc/sh/rcar/dma.c | 32 +++++ sound/soc/sh/rcar/rsnd.h | 5 + sound/soc/sh/rcar/ssi.c | 56 +++++++-- 8 files changed, 451 insertions(+), 69 deletions(-) -- 2.17.0

3 6

[alsa-devel] i2s\pcm\spdif\slimbus
by Muni Sekhar 29 Sep '18

29 Sep '18

Hi All, Can someone share the links to i2s\pcm\spdif\slimbus bus specifications. -- Thanks, Sekhar

1 0

[alsa-devel] Fwd: Help with sound on Cherry Trail laptop with ES8316 sound chip
by Sundar Nagarajan 29 Sep '18

29 Sep '18

Hello, I have (separate) issues with two different Cherry Trail laptops that have the ES8316 chipset for sound. I already posted the following issues on Pierre Bossart's Intel UCM repository(https://github.com/plbossart/UCM): Issue 42: RDP Thinbook 1430 laptop (ES8316): Can't select between built-in speakers and headphone output (https://github.com/plbossart/UCM/issues/42) Pierre's input was as follows: The UCM file you are using gives the clue already: #Codec IC has only 1 analog out, switching SPK/HP is done externally, #probably by means of some analog switch controlled by gpio There is nothing we can do at the UCM level, this has to be handled at a lower level (i.e. the kernel) In addition, on a different laptop, I have a different problem with the ES3816 chipset. Issue 41: No sound on speakers on Chuwi Hi12 laptop - headphone sound works (es8316) (https://github.com/plbossart/UCM/issues/41) I reached out to Mark Brown, Daniel Drake and David Yang who were listed on the kernel.org commit for the ytcht_es8316.c driver (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…) and Mark suggested reaching out to you folks. What is the right place for me to post my issues to get your ideas and help? I have access to these machines, and I can compile kernels and perform testing, collect logs etc. Regards, Sundar ---------- Forwarded message --------- From: Mark Brown <broonie(a)kernel.org> Date: Thu, Sep 27, 2018 at 3:14 PM Subject: Re: Help with sound on Cherry Trail laptop with ES8316 sound chip To: Sundar Nagarajan <sun.nagarajan(a)gmail.com> Cc: Daniel Drake <drake(a)endlessm.com>, David Yang <yangxiaohua(a)everest-semi.com> On Wed, Sep 26, 2018 at 12:53:44PM -0700, Sundar Nagarajan wrote: > )I found your email from the kernel.org commit log for the > bytcht_es8316.c driver > (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…. > I have (separate) issues with two different Cherry Trail laptops that > have the ES8316 chipset. I posted the following issues on Pierre > Bossart's Intel UCM repository(https://github.com/plbossart/UCM): He's on sabbatical at the minute FWIW. > What is the right place for me to post my issues to get your ideas and help? You'd be best contacting the Intel maintainers about x86 specific issues: INTEL ASoC DRIVERS M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> M: Jie Yang <yang.jie(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ That includes Pierre but also a few other people, hopefully at least one of them will be able to help or point you to someone who can. You should also copy in the alsa-devel list (alsa-devel(a)alsa-project.org) in case someone else can help and so people can google the discussion later, though that'll mean your e-mail address ends up in the archives so you might not be happy with that.

1 0

[alsa-devel] [PATCH v2] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops
by Matthias Kaehlcke 28 Sep '18

28 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - use 'snd_dummy_dma_ops' instead of 'snd_dummy_ops' --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..e3b9dd634c6d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_dma_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

2 1

[alsa-devel] [PATCH] ASoC: pxa: avoid AC97_BUS build warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting AC97_BUS_NEW from SND_PXA2XX_SOC_AC97 leads to a Kconfig warning if any other driver selects AC97_BUS: WARNING: unmet direct dependencies detected for AC97_BUS_COMPAT Depends on [n]: SOUND [=y] && !UML && SND [=y] && AC97_BUS_NEW [=y] && !AC97_BUS [=y] Selected by [y]: - SND_SOC_WM9713 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && AC97_BUS_NEW [=y] I don't know if that combination is supposed to work. Assuming it is not, this adds a dependency on all users for PXA to avoids the combination. Fixes: 1c8bc7b3de5e ("ASoC: pxa: switch to new ac97 bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 29f577e6dfc0..943b44de1464 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -88,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,6 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,6 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 + depends on !AC97_BUS select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -116,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -126,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,6 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -194,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- 2.18.0

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[alsa-devel] [PATCH] ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting SND_SOC_WM8731 is only allowed when all its dependencies are already there: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] Selected by [m]: - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && (ARCH_AT91 [=y] || COMPILE_TEST [=y]) && ATMEL_SSC [=y] && SND_SOC_I2C_AND_SPI [=m] - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && COMPILE_TEST [=y] && SND_SOC_I2C_AND_SPI [=m] Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 81a0712d4f14..64f86f0b87e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -101,6 +101,7 @@ config SND_ATMEL_SOC_I2S config SND_SOC_MIKROE_PROTO tristate "Support for Mikroe-PROTO board" depends on OF + depends on SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 help Say Y or M if you want to add support for MikroElektronika PROTO Audio -- 2.18.0

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