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[sound:test/usb-hacks 14/20] sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used
by kernel test robot 10 Nov '20

10 Nov '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-hacks head: 2adedc89c82670019ee3b016d2a2237176768e3a commit: 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 [14/20] ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list config: i386-allyesconfig (attached as .config) compiler: gcc-9 (Debian 9.3.0-15) 9.3.0 reproduce (this is a W=1 build): # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-hacks git checkout 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 # save the attached .config to linux build tree make W=1 ARCH=i386 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): sound/usb/pcm.c: In function 'set_format': >> sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used [-Wunused-but-set-variable] 579 | struct usb_interface_descriptor *altsd; | ^~~~~ vim +/altsd +579 sound/usb/pcm.c 71bb64c56d787a Eldad Zack 2013-08-03 571 71bb64c56d787a Eldad Zack 2013-08-03 572 /* 71bb64c56d787a Eldad Zack 2013-08-03 573 * find a matching format and set up the interface 71bb64c56d787a Eldad Zack 2013-08-03 574 */ 71bb64c56d787a Eldad Zack 2013-08-03 575 static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) 71bb64c56d787a Eldad Zack 2013-08-03 576 { 71bb64c56d787a Eldad Zack 2013-08-03 577 struct usb_device *dev = subs->dev; 71bb64c56d787a Eldad Zack 2013-08-03 578 struct usb_host_interface *alts; 71bb64c56d787a Eldad Zack 2013-08-03 @579 struct usb_interface_descriptor *altsd; 71bb64c56d787a Eldad Zack 2013-08-03 580 struct usb_interface *iface; 71bb64c56d787a Eldad Zack 2013-08-03 581 int err; 71bb64c56d787a Eldad Zack 2013-08-03 582 71bb64c56d787a Eldad Zack 2013-08-03 583 iface = usb_ifnum_to_if(dev, fmt->iface); 71bb64c56d787a Eldad Zack 2013-08-03 584 if (WARN_ON(!iface)) 71bb64c56d787a Eldad Zack 2013-08-03 585 return -EINVAL; b099b9693d23d0 Takashi Iwai 2018-05-02 586 alts = usb_altnum_to_altsetting(iface, fmt->altsetting); 0141254b0a74b3 Johan Hovold 2019-12-20 587 if (WARN_ON(!alts)) 71bb64c56d787a Eldad Zack 2013-08-03 588 return -EINVAL; 0141254b0a74b3 Johan Hovold 2019-12-20 589 altsd = get_iface_desc(alts); 71bb64c56d787a Eldad Zack 2013-08-03 590 92adc96f8eecd9 Hui Wang 2019-12-18 591 if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) 71bb64c56d787a Eldad Zack 2013-08-03 592 return 0; 71bb64c56d787a Eldad Zack 2013-08-03 593 71bb64c56d787a Eldad Zack 2013-08-03 594 /* close the old interface */ 92adc96f8eecd9 Hui Wang 2019-12-18 595 if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { 8a463225b11047 Takashi Iwai 2018-05-02 596 if (!subs->stream->chip->keep_iface) { 71bb64c56d787a Eldad Zack 2013-08-03 597 err = usb_set_interface(subs->dev, subs->interface, 0); 71bb64c56d787a Eldad Zack 2013-08-03 598 if (err < 0) { 0ba41d917eeb87 Takashi Iwai 2014-02-26 599 dev_err(&dev->dev, 0ba41d917eeb87 Takashi Iwai 2014-02-26 600 "%d:%d: return to setting 0 failed (%d)\n", 0ba41d917eeb87 Takashi Iwai 2014-02-26 601 fmt->iface, fmt->altsetting, err); 71bb64c56d787a Eldad Zack 2013-08-03 602 return -EIO; 71bb64c56d787a Eldad Zack 2013-08-03 603 } 8a463225b11047 Takashi Iwai 2018-05-02 604 } 71bb64c56d787a Eldad Zack 2013-08-03 605 subs->interface = -1; 71bb64c56d787a Eldad Zack 2013-08-03 606 subs->altset_idx = 0; 71bb64c56d787a Eldad Zack 2013-08-03 607 } 71bb64c56d787a Eldad Zack 2013-08-03 608 92adc96f8eecd9 Hui Wang 2019-12-18 609 if (subs->need_setup_fmt) 92adc96f8eecd9 Hui Wang 2019-12-18 610 subs->need_setup_fmt = false; 92adc96f8eecd9 Hui Wang 2019-12-18 611 71bb64c56d787a Eldad Zack 2013-08-03 612 /* set interface */ b099b9693d23d0 Takashi Iwai 2018-05-02 613 if (iface->cur_altsetting != alts) { 6874daad4b0fbe Jurgen Kramer 2014-11-28 614 err = snd_usb_select_mode_quirk(subs, fmt); 6874daad4b0fbe Jurgen Kramer 2014-11-28 615 if (err < 0) 6874daad4b0fbe Jurgen Kramer 2014-11-28 616 return -EIO; 6874daad4b0fbe Jurgen Kramer 2014-11-28 617 71bb64c56d787a Eldad Zack 2013-08-03 618 err = usb_set_interface(dev, fmt->iface, fmt->altsetting); 71bb64c56d787a Eldad Zack 2013-08-03 619 if (err < 0) { 0ba41d917eeb87 Takashi Iwai 2014-02-26 620 dev_err(&dev->dev, 0ba41d917eeb87 Takashi Iwai 2014-02-26 621 "%d:%d: usb_set_interface failed (%d)\n", 0ba41d917eeb87 Takashi Iwai 2014-02-26 622 fmt->iface, fmt->altsetting, err); 71bb64c56d787a Eldad Zack 2013-08-03 623 return -EIO; 71bb64c56d787a Eldad Zack 2013-08-03 624 } 0ba41d917eeb87 Takashi Iwai 2014-02-26 625 dev_dbg(&dev->dev, "setting usb interface %d:%d\n", 71bb64c56d787a Eldad Zack 2013-08-03 626 fmt->iface, fmt->altsetting); 71bb64c56d787a Eldad Zack 2013-08-03 627 snd_usb_set_interface_quirk(dev); 71bb64c56d787a Eldad Zack 2013-08-03 628 } 71bb64c56d787a Eldad Zack 2013-08-03 629 b099b9693d23d0 Takashi Iwai 2018-05-02 630 subs->interface = fmt->iface; b099b9693d23d0 Takashi Iwai 2018-05-02 631 subs->altset_idx = fmt->altset_idx; 71bb64c56d787a Eldad Zack 2013-08-03 632 subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, 71bb64c56d787a Eldad Zack 2013-08-03 633 alts, fmt->endpoint, subs->direction, 71bb64c56d787a Eldad Zack 2013-08-03 634 SND_USB_ENDPOINT_TYPE_DATA); 71bb64c56d787a Eldad Zack 2013-08-03 635 71bb64c56d787a Eldad Zack 2013-08-03 636 if (!subs->data_endpoint) 71bb64c56d787a Eldad Zack 2013-08-03 637 return -EINVAL; 71bb64c56d787a Eldad Zack 2013-08-03 638 7502ae0cddc65a Takashi Iwai 2020-11-05 639 err = set_sync_endpoint(subs, fmt); 71bb64c56d787a Eldad Zack 2013-08-03 640 if (err < 0) 71bb64c56d787a Eldad Zack 2013-08-03 641 return err; 71bb64c56d787a Eldad Zack 2013-08-03 642 d133f2c22e9cb7 Eldad Zack 2013-08-03 643 err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); d133f2c22e9cb7 Eldad Zack 2013-08-03 644 if (err < 0) e5779998bf8b70 Daniel Mack 2010-03-04 645 return err; e5779998bf8b70 Daniel Mack 2010-03-04 646 e5779998bf8b70 Daniel Mack 2010-03-04 647 subs->cur_audiofmt = fmt; e5779998bf8b70 Daniel Mack 2010-03-04 648 e5779998bf8b70 Daniel Mack 2010-03-04 649 snd_usb_set_format_quirk(subs, fmt); e5779998bf8b70 Daniel Mack 2010-03-04 650 e5779998bf8b70 Daniel Mack 2010-03-04 651 return 0; e5779998bf8b70 Daniel Mack 2010-03-04 652 } e5779998bf8b70 Daniel Mack 2010-03-04 653 :::::: The code at line 579 was first introduced by commit :::::: 71bb64c56d787a221752b1de034fe8c07c737f5c ALSA: usb-audio: separate sync endpoint setting from set_format :::::: TO: Eldad Zack <eldad(a)fogrefinery.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH AUTOSEL 4.19 02/21] ALSA: hda: Reinstate runtime_allow() for all hda controllers
by Sasha Levin 10 Nov '20

10 Nov '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 9fc149c3bce7bdbb94948a8e6bd025e3b3538603 ] The broken jack detection should be fixed by commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), let's try enabling runtime PM by default again. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20201027130038.16463-4-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d43245937db7e..8e1eb5f243a27 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2478,6 +2478,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.27.0

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[PATCH AUTOSEL 5.4 01/42] ASoC: qcom: sdm845: set driver name correctly
by Sasha Levin 10 Nov '20

10 Nov '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 3f48b6eba15ea342ef4cb420b580f5ed6605669f ] With the current state of code, we would endup with something like below in /proc/asound/cards for 2 machines based on this driver. Machine 1: 0 [DB845c ]: DB845c - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: Lenovo-YOGA-C63 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 This is not very UCM friendly both w.r.t to common up configs and card identification, and UCM2 became totally not usefull with just one ucm sdm845.conf for two machines which have different setups w.r.t HDMI and other dais. Reasons for such thing is partly because Qualcomm machine drivers never cared to set driver_name. This patch sets up driver name for the this driver to sort out the UCM integration issues! after this patch contents of /proc/asound/cards: Machine 1: 0 [DB845c ]: sdm845 - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: sdm845 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 with this its possible to align with what UCM2 expects and we can have sdm845/DB845.conf sdm845/LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216.conf ... for board variants. This should scale much better! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20201023095849.22894-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 7e6c41e63d8e1..23e1de61e92e4 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -16,6 +16,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -407,6 +408,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; -- 2.27.0

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[PATCH AUTOSEL 5.9 01/55] ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015
by Sasha Levin 10 Nov '20

10 Nov '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit eb5a558705c7f63d06b4ddd072898b1ca894e053 ] RT1015's output widget name is "SPO" instead of "Speaker". Fixes it to use the correct names. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20201019044724.1601476-1-tzungbi@google.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 31 +++++++++++++++---- 1 file changed, 25 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index a6c690c5308d3..58b76e985f7f3 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -624,15 +624,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = { + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_rt1015_card = { .name = "mt8183_da7219_rt1015", .owner = THIS_MODULE, - .controls = mt8183_da7219_max98357_snd_controls, - .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), - .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), - .dapm_routes = mt8183_da7219_max98357_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), + .controls = mt8183_da7219_rt1015_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls), + .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets), + .dapm_routes = mt8183_da7219_rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes), .dai_link = mt8183_da7219_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, -- 2.27.0

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ASoC: Question regarding device-tree multi-lane I2S for Allwinner SoC
by Clément Péron 09 Nov '20

09 Nov '20

Hi device-tree and sound Maintainers, I have a question regarding multi-lane i2S representation. On the Allwinner SoC the I2S/PCM Interface can handle up to four lanes as output or input. For each lane we can enable up to sixteen slots. And for each output slot we can choose which slot to map. The only representation I found is for Amlogic device-tree they did the following sound node : /* 8ch hdmi interface */ dai-link-7 { sound-dai = <&tdmif_b>; dai-format = "i2s"; dai-tdm-slot-tx-mask-0 = <1 1>; dai-tdm-slot-tx-mask-1 = <1 1>; dai-tdm-slot-tx-mask-2 = <1 1>; dai-tdm-slot-tx-mask-3 = <1 1>; mclk-fs = <256>; codec { sound-dai = <&tohdmitx TOHDMITX_I2S_IN_B>; }; }; This kind of representation gives the information that 2 slots should be enabled as TX per lane but don't give which slot to map. I was thinking about a representation per lane but maybe it's a bit complicated ? dai-format = "dsp_a"; dai-tdm-slot-width = <32>; // Lane 0 : Output 8 channels 0-7 using TDM dai-tdm-slot-tx-mask-0 = <1 1 1 1 1 1 1 1>; dai-tdm-slot-chmap-0 = <0 1 2 3 4 5 6 7>; // Lane 1 : Output 3 channels 5-7 using TDM dai-tdm-slot-tx-mask-1 = <1 1 1>; dai-tdm-slot-chmap-1 = <5 6 7>; I will only start to support HDMI, so 4 I2S lanes for now it should look like this dai-format = "i2s"; dai-tdm-slot-width = <32>; frame-inversion; // Lane 0 : Output channels 0,1 dai-tdm-slot-tx-mask-0 = <1 1>; dai-tdm-slot-chmap-0 = <0 1>; // Lane 1: Output channels 2,3 dai-tdm-slot-tx-mask-1 = <1 1>; dai-tdm-slot-chmap-1 = <2 3>; // Lane 2: Output channels 4,5 dai-tdm-slot-tx-mask-2 = <1 1>; dai-tdm-slot-chmap-2 = <4 5>; // Lane 3: Output channels 6,7 dai-tdm-slot-tx-mask-3 = <1 1>; dai-tdm-slot-chmap-3 = <6 7>; What do you think? Do you have any remark / idea about this ? Thanks for your help Clement

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USB Audio device: Shure MV7 not recognized
by GitHub issues - edited 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was edited from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=7bfa1d9ac09aac53a52834e95b833d299dfe3896 `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. The device has a head-phone jack with feedback from the mic, and without pulseaudio running I hear the microphone inputs. So I was wondering if this was in particular a pulseaudio issue. ``` > arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic [HD-Audio Generic], device 0: ALCS1200A Analog [ALCS1200A Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 2: ALCS1200A Alt Analog [ALCS1200A Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: C925e [Logitech Webcam C925e], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: MV7 [Shure MV7], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 > arecord -d 3 -f U8 sample.mp3 ALSA lib pcm_dsnoop.c:638:(snd_pcm_dsnoop_open) unable to open slave arecord: main:830: audio open error: No such file or directory ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

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USB Audio device: Shure MV7 not recognized
by GitHub issues - edited 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was edited from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=7bfa1d9ac09aac53a52834e95b833d299dfe3896 `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

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USB Audio device: Shure MV7 not recognized
by GitHub issues - opened 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was opened from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=1fb51d03454930fce4d6164323cfb7b3c7c88dda `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: qcom: sc7180: Add missing PM ops
by Cheng-Yi Chiang 09 Nov '20

09 Nov '20

Use PM ops snd_soc_pm_ops to handle suspend/resume like other machine drivers. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sc7180.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index b391f64c3a80..42e366ecc689 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -258,6 +258,7 @@ static struct platform_driver sc7180_snd_driver = { .driver = { .name = "msm-snd-sc7180", .of_match_table = sc7180_snd_device_id, + .pm = &snd_soc_pm_ops, }, }; module_platform_driver(sc7180_snd_driver); -- 2.29.2.222.g5d2a92d10f8-goog

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[PATCH 0/4] ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully
by Peter Ujfalusi 09 Nov '20

09 Nov '20

Hi, The series is inspired by the effort to standardize TI's arm64 dtsi files to keep all nodes in 'okay' state and let the board dts files disable not needed peripherals (and not properly configured): https://lore.kernel.org/lkml/20201104224356.18040-1-nm@ti.com/ In the unlikely (or likely?) event when the dts misses to disable the McASP node which is not configured we currenly and luckily just manage to not crash as we had fixup code in place for legacy pdata boots. This however prints out a message which does not really help to identify the issue. This series will reduce some of the noise during boot (first patch) then adds the needed changes to handle the variations of 'okay': A - have all required DT properties for audio B - gpiochip is enabled A && !B -> everything is OK for audio, no gpiochip registered A && B -> everything is OK for audio, gpiochip is registered !A && B -> audio is not OK, gpiochip is registered. dev_dbg() for audio and do not register SOC DAI and PCM !A && !B -> audio is not OK, no gpiochip. dev_err() and fail the probe Regards, Peter --- Peter Ujfalusi (4): ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling ASoC: ti: davinci-mcasp: Handle missing required DT properties sound/soc/ti/davinci-mcasp.c | 294 ++++++++++++++--------------------- 1 file changed, 119 insertions(+), 175 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 5

[PATCH] ASoC: rt1015: modification for calibration to get better performance
by jack.yu＠realtek.com 09 Nov '20

09 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Modification for calibration to get better performance. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++-------------------- 1 file changed, 48 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 52787fbd522f..32d748cc5ae3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -497,18 +497,40 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) snd_soc_dapm_mutex_lock(&component->dapm); regcache_cache_bypass(regmap, true); - regmap_write(regmap, RT1015_PWR1, 0xd7df); - regmap_write(regmap, RT1015_PWR4, 0x00b2); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_PWR9, 0xAA60); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0089); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008A); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008C); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008D); + regmap_write(regmap, RT1015_PWR4, 0x80B2); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x5797); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2100); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0100); + regmap_write(regmap, RT1015_PWR5, 0x2175); + regmap_write(regmap, RT1015_MIXER1, 0x005D); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12F7); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x1205); + msleep(200); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2000); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0180); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x0A05); + msleep(200); + regmap_write(regmap, RT1015_PWR4, 0x00B2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2028); regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); - regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); - msleep(500); - regmap_write(regmap, RT1015_PWR1, 0x0); + regmap_write(regmap, RT1015_PWR5, 0x0175); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x1721); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x570E); + regmap_write(regmap, RT1015_MIXER1, 0x203D); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5A01); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12FF); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0eFE); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008E); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0088); + regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); + regmap_write(regmap, RT1015_SYS_RST2, 0x0b9a); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); @@ -603,6 +625,8 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b0a); snd_soc_component_write(component, RT1015_GAT_BOOST, 0xacfe); snd_soc_component_write(component, @@ -610,10 +634,14 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, RT1015_GAT_BOOST, 0xecfe); } else { + snd_soc_component_write(component, + 0x032d, 0xaa60); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x026e); + RT1015_SYS_RST2, 0x0b0a); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x008e); } break; @@ -627,11 +655,17 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_PWR9, 0xa800); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } else { snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x0268); + 0x032d, 0xaa60); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x0088); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } rt1015->dac_is_used = 0; @@ -664,38 +698,12 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("LDO2", RT1015_PWR1, RT1015_PWR_LDO2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("INT RC CLK", RT1015_PWR1, RT1015_PWR_INTCLK_BIT, - 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ISENSE", RT1015_PWR1, RT1015_PWR_ISENSE_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VSENSE", RT1015_PWR1, RT1015_PWR_VSENSE_BIT, 0, - NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", RT1015_PWR1, RT1015_PWR_PLL_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("BG1 BG2", RT1015_PWR1, RT1015_PWR_BG_1_2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("MBIAS BG", RT1015_PWR1, RT1015_PWR_MBIAS_BG_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VBAT", RT1015_PWR1, RT1015_PWR_VBAT_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MBIAS", RT1015_PWR1, RT1015_PWR_MBIAS_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("ADCV", RT1015_PWR1, RT1015_PWR_ADCV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MIXERV", RT1015_PWR1, RT1015_PWR_MIXERV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("SUMV", RT1015_PWR1, RT1015_PWR_SUMV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("VREFLV", RT1015_PWR1, RT1015_PWR_VREFLV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), @@ -703,19 +711,7 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { static const struct snd_soc_dapm_route rt1015_dapm_routes[] = { { "DAC", NULL, "AIFRX" }, - { "DAC", NULL, "LDO2" }, { "DAC", NULL, "PLL", rt1015_is_sys_clk_from_pll}, - { "DAC", NULL, "INT RC CLK" }, - { "DAC", NULL, "ISENSE" }, - { "DAC", NULL, "VSENSE" }, - { "DAC", NULL, "BG1 BG2" }, - { "DAC", NULL, "MBIAS BG" }, - { "DAC", NULL, "VBAT" }, - { "DAC", NULL, "MBIAS" }, - { "DAC", NULL, "ADCV" }, - { "DAC", NULL, "MIXERV" }, - { "DAC", NULL, "SUMV" }, - { "DAC", NULL, "VREFLV" }, { "Amp Drv", NULL, "DAC" }, { "SPO", NULL, "Amp Drv" }, }; @@ -1058,7 +1054,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; rt1015->cali_done = 0; - snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); -- 2.29.0

2 1

[PATCH] ASoC: fsl_aud2htx: Remove dev_err() usage after platform_get_irq()
by Shengjiu Wang 09 Nov '20

09 Nov '20

platform_get_irq() would print error message internally, so dev_err() after platform_get_irq() is not needed Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 124aeb70f24e..4091ccc7c3e9 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -211,11 +211,8 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", - dev_name(&pdev->dev)); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, fsl_aud2htx_isr, 0, dev_name(&pdev->dev), aud2htx); -- 2.27.0

3 2

[PATCH v1 0/2] Support SC7180 sound card variation without headset
by xuyuqing 09 Nov '20

09 Nov '20

This series is based on patch [v5,2/2] ASoC: qcom: sc7180: Modify machine driver for 2mic Use compatible string to let machine driver handle the case where ALC5682 headset codec is replaced with ADAU7002 dmic codec xuyuqing (2): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Modify machine driver for adau7002 .../bindings/sound/google,sc7180-trogdor.yaml | 4 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 119 ++++++++++++++++-- 3 files changed, 112 insertions(+), 12 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 09 Nov '20

09 Nov '20

[ Upstream commit 1b290ef023b3eeb4f4688b582fecb773915ef937 ] Add alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. This change addresses multiple issues with current mechanism, for example - there are devices without NHLT table, and that currently results in tplg_name being empty. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200427132727.24942-2-mateusz.gorski@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- This functionality is merged on upstream kernel and widely used. Merging it to LTS kernel would improve the user experience and resolve some of the problems regarding topology naming that the users are facing. sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..4b114ece58c6 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -14,6 +14,7 @@ #include <linux/uuid.h> #include <sound/intel-nhlt.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> #include <uapi/sound/skl-tplg-interface.h> @@ -3565,8 +3566,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->mach->drv_name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3588,8 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

4 6

[PATCH] ALSA: usb-audio: Fix coding style problems
by Atul 08 Nov '20

08 Nov '20

From: Atul Gopinathan <leoatul12(a)gmail.com> Hi, This patch fixes some coding style warnings and errors that were generated using scripts/checkpatch.pl. Signed-off-by: Atul Gopinathan <leoatul12(a)gmail.com> --- sound/usb/pcm.c | 16 ++++++++-------- sound/usb/quirks.c | 21 +++++++++++---------- 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a860303cc522..ed5c77ce8e14 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -104,7 +104,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs) continue; } attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (! found) { + if (!found) { found = fp; cur_attr = attr; continue; @@ -304,7 +304,7 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, if (altsd->bAlternateSetting != altsetting || altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC || (altsd->bInterfaceSubClass != 2 && - altsd->bInterfaceProtocol != 2 ) || + altsd->bInterfaceProtocol != 2) || altsd->bNumEndpoints < 1) return -ENOENT; epd = get_endpoint(*alts, 0); @@ -482,8 +482,9 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ + * because there is a version of the SB Audigy 2 NX firmware lacking + * the audio fields in the endpoint descriptors + */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0)) { @@ -911,7 +912,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) struct usb_interface *iface; int ret; - if (! subs->cur_audiofmt) { + if (!subs->cur_audiofmt) { dev_err(&subs->dev->dev, "no format is specified!\n"); return -ENXIO; } @@ -973,9 +974,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return ret; } -static const struct snd_pcm_hardware snd_usb_hardware = -{ - .info = SNDRV_PCM_INFO_MMAP | +static const struct snd_pcm_hardware snd_usb_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_INTERLEAVED | diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c989ad8052ae..b6f3345db93f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -398,7 +398,7 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -593,7 +593,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) { dev_dbg(&dev->dev, "sending Extigy boot sequence...\n"); /* Send message to force it to reconnect with full interface. */ - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x10, 0x43, 0x0001, 0x000a, NULL, 0); if (err < 0) dev_dbg(&dev->dev, "error sending boot message: %d\n", err); @@ -646,8 +646,9 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) "error usb_driver_set_configuration: %d\n", err); /* Always return an error, so that we stop creating a device - that will just be destroyed and recreated with a new - configuration */ + * that will just be destroyed and recreated with a new + * configuration + */ return -ENODEV; } else dev_info(&dev->dev, "Fast Track Pro config OK\n"); @@ -895,9 +896,10 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) usb_reset_device(dev); /* return -EAGAIN, so the creation of an audio interface for this - * temporary device is aborted. The device will reconnect with a - * new product ID */ - return -EAGAIN; + * temporary device is aborted. The device will reconnect with a + * new product ID + */ + return -EAGAIN; } static void mbox2_setup_48_24_magic(struct usb_device *dev) @@ -919,7 +921,6 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev) 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); - return; } /* Digidesign Mbox 2 needs to load firmware onboard @@ -1830,8 +1831,8 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is an older model 77d:223) */ -- 2.25.1

3 2

ThinkPad support headset button
by Kailang 07 Nov '20

07 Nov '20

Hi Takashi, Two ThinkPad want to support headset button. Attach was support patch. Many Thanks. BR, Kailang

2 1

Re: [Sound-open-firmware] linux-next: Tree for Nov 6 (sound/soc/sof/sof-pci-dev.o)
by Pierre-Louis Bossart 06 Nov '20

06 Nov '20

> on x86_64: > > ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': > sof-pci-dev.c:(.text+0x5c): undefined reference to `snd_intel_dsp_driver_probe' > > > Full randconfig file is attached. Nice catch, thanks Randy! Looks like we put the select SND_INTEL_DSP_CONFIG in the wrong place, it's not dependent on the HDaudio link being selected. Clearly a bug we've had for a while. The diff below makes the error go away but I have to run it past folks who are already enjoying their week-end. Will follow-up next week with a proper fix. diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index a066e08860cb..5bfc2f8b13b9 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -271,6 +271,7 @@ config SND_SOC_SOF_JASPERLAKE config SND_SOC_SOF_HDA_COMMON tristate + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE help @@ -330,7 +331,6 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level

2 1

[PATCH v3 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 06 Nov '20

06 Nov '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3: - Add additionalProperties changes in v2: - fix indentation issue - remove nodename .../bindings/sound/fsl,aud2htx.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..aa4be7170717 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

5 8

[PATCH 0/7] PinePhone BT audio bringup
by Samuel Holland 06 Nov '20

06 Nov '20

This series makes use of the additional DAIs recently added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver similar to bt-sco, and I'm not sure if such a driver would be desired upstream. The modem has a similar amount of configurability as the rtl8723cs BT chip. For the BT chip, the DAI format and PCM parameters are set in a configuration firmware file and loaded at driver load time. For the modem, the DAI format and PCM parameters are set by (and can be queried from) an AT command. However, this AT command requires a modem restart to take effect, so the parameters cannot feasibly be changed at runtime. With a dummy driver, we pick some "standard" set of PCM parameters, e.g. 16 bit mono at 8 or 16 kHz, and assume the hardware agrees. Similarly, we assume the DAI format pre-programmed in the remote hardware matches what is configured in the DAI link (in this case, in the device tree). Is that the right way to model this? Or does the modem (and maybe even the BT chip) deserve a more featureful driver that reads and/or programs the format and params at probe time? Because of those unanswered questions, I'm sending BT SCO support first/only. Regards, Samuel Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 2 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 55 insertions(+), 13 deletions(-) -- 2.26.2

3 9

[PATCH v3] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 06 Nov '20

06 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 ++++++++++++++ sound/soc/codecs/rt1015.c | 20 +++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 4 files changed, 43 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt index fcfd02d8d32f..e498966d436f 100644 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -8,10 +8,16 @@ Required properties: - reg : The I2C address of the device. +Optional properties: + +- realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. Example: rt1015: codec@28 { compatible = "realtek,rt1015"; reg = <0x28>; + realtek,power-up-delay-ms = <50>; }; diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..70a7538d4c89 --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay_ms; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..967193518349 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay_ms = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay_ms); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay-ms", + &rt1015->pdata.power_up_delay_ms); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

3 2

[PATCH v6 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 06 Nov '20

06 Nov '20

Hi Rob, Mark Cc: Sameer These are v6 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v5 -> v6 - add Acked-by/Reviewed-by on each patch - remove "audio-graph-card,xxx" property - remove un-needed if "ok" v4 -> v5 - move "endpoint" under "port" v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 153 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 451 +++++++++++++++ 4 files changed, 604 insertions(+), 857 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

3 5

[PATCH mlx5-next v1 00/11] Convert mlx5 to use auxiliary bus
by Leon Romanovsky 06 Nov '20

06 Nov '20

From: Leon Romanovsky <leonro(a)nvidia.com> Changelog: v1: * Renamed _mlx5_rescan_driver to be mlx5_rescan_driver_locked like in other parts of the mlx5 driver. * Renamed MLX5_INTERFACE_PROTOCOL_VDPA to tbe MLX5_INTERFACE_PROTOCOL_VNET as a preparation to coming series from Eli C. * Some small naming renames in mlx5_vdpa. * Refactored adev index code to make Parav's SF series to apply more easily. * Fixed devlink reload bug that caused to lost TCP connection. v0: https://lore.kernel.org/lkml/20201026111849.1035786-1-leon@kernel.org/ -------------------------------------------------------------- Hi, This patch set converts mlx5 driver to use auxiliary bus [1]. In this series, we are connecting three subsystems (VDPA, netdev and RDMA) through mlx5_core PCI driver. That driver is responsible to create proper devices based on supported firmware. First four patches are preparitions and fixes that were spotted during code development, rest is the conversion itself. Thanks [1] https://lore.kernel.org/lkml/20201023003338.1285642-1-david.m.ertman@intel.… Leon Romanovsky (11): net/mlx5: Don't skip vport check net/mlx5: Properly convey driver version to firmware net/mlx5_core: Clean driver version and name vdpa/mlx5: Make hardware definitions visible to all mlx5 devices net/mlx5: Register mlx5 devices to auxiliary virtual bus vdpa/mlx5: Connect mlx5_vdpa to auxiliary bus net/mlx5e: Connect ethernet part to auxiliary bus RDMA/mlx5: Convert mlx5_ib to use auxiliary bus net/mlx5: Delete custom device management logic net/mlx5: Simplify eswitch mode check RDMA/mlx5: Remove IB representors dead code drivers/infiniband/hw/mlx5/counters.c | 7 - drivers/infiniband/hw/mlx5/ib_rep.c | 113 ++-- drivers/infiniband/hw/mlx5/ib_rep.h | 45 +- drivers/infiniband/hw/mlx5/main.c | 148 +++-- drivers/infiniband/hw/mlx5/mlx5_ib.h | 4 +- .../net/ethernet/mellanox/mlx5/core/Kconfig | 1 + drivers/net/ethernet/mellanox/mlx5/core/dev.c | 567 ++++++++++++------ .../net/ethernet/mellanox/mlx5/core/devlink.c | 4 +- .../ethernet/mellanox/mlx5/core/en_ethtool.c | 4 +- .../net/ethernet/mellanox/mlx5/core/en_main.c | 135 ++--- .../net/ethernet/mellanox/mlx5/core/en_rep.c | 42 +- .../net/ethernet/mellanox/mlx5/core/en_rep.h | 6 +- .../net/ethernet/mellanox/mlx5/core/en_tc.c | 8 +- .../mellanox/mlx5/core/esw/devlink_port.c | 2 +- .../net/ethernet/mellanox/mlx5/core/eswitch.c | 28 +- .../mellanox/mlx5/core/eswitch_offloads.c | 6 + .../mellanox/mlx5/core/ipoib/ethtool.c | 2 +- drivers/net/ethernet/mellanox/mlx5/core/lag.c | 58 +- .../net/ethernet/mellanox/mlx5/core/main.c | 50 +- .../ethernet/mellanox/mlx5/core/mlx5_core.h | 33 +- drivers/vdpa/mlx5/Makefile | 2 +- drivers/vdpa/mlx5/net/main.c | 76 --- drivers/vdpa/mlx5/net/mlx5_vnet.c | 53 +- drivers/vdpa/mlx5/net/mlx5_vnet.h | 24 - include/linux/mlx5/driver.h | 34 +- include/linux/mlx5/eswitch.h | 8 +- .../linux/mlx5/mlx5_ifc_vdpa.h | 6 +- 27 files changed, 818 insertions(+), 648 deletions(-) delete mode 100644 drivers/vdpa/mlx5/net/main.c delete mode 100644 drivers/vdpa/mlx5/net/mlx5_vnet.h rename drivers/vdpa/mlx5/core/mlx5_vdpa_ifc.h => include/linux/mlx5/mlx5_ifc_vdpa.h (97%) -- 2.28.0

8 28

[PATCH v5 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 06 Nov '20

06 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> v5: - Machine driver: - Fix a format string warning (Reported-by: kernel test robot <lkp(a)intel.com>). detailed info at https://lore.kernel.org/patchwork/patch/1331087/ v4: - Machine driver: Addressed suggestions from Tzung-Bi. - Remove redundant judgments in dmic_set() and dmic_get(). - Remove 1 level indent of judgment of IS_ERR(data->dmic_sel). v3: - Machine driver: Addressed suggestions from Tzung-Bi. - move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data. - Remove redundant judgments in dmic_set(). v2: - Documentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 58 ++++++++++++++++++ sound/soc/qcom/sc7180.c | 61 +++++++++++++++++++ 2 files changed, 119 insertions(+) -- 2.25.1

3 5

[PATCH AUTOSEL 5.9 039/147] ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
by Sasha Levin 05 Nov '20

05 Nov '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 6c63c954e1c52f1262f986f36d95f557c6f8fa94 ] When hda_codec_probe() doesn't initialize audio component, we disable the codec and keep going. However,the resources are not released. The child_count of SOF device is increased in snd_hdac_ext_bus_device_init but is not decrease in error case, so SOF can't get suspended. snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it gets a error. Now copy this behavior to release resources and decrease SOF device child_count to release SOF device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d7..c475955c6eeba 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -151,7 +151,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - return -ENOENT; + goto error; } hda_priv->need_display_power = true; } @@ -174,7 +174,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - ret = -ENOENT; + goto error; } return ret; -- 2.25.1

3 3

[GIT PULL] ASoC fixes for v5.10-rc2
by Mark Brown 05 Nov '20

05 Nov '20

The following changes since commit 3650b228f83adda7e5ee532e2b90429c03f7b9ec: Linux 5.10-rc1 (2020-10-25 15:14:11 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.10-rc2 for you to fetch changes up to f9d7c6eb23f7e55e7a0ca5451da06909bdfdd0e4: ASoC: mchp-spdiftx: Do not set Validity bit(s) (2020-11-04 17:50:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.10 A batch of driver specific fixes that have come up since the merge window, nothing particularly major here but all good to have. ---------------------------------------------------------------- Bard Liao (1): ASoC: SOF: loader: handle all SOF_IPC_EXT types Cezary Rojewski (3): ASoC: Intel: atom: Remove duplicate kconfigs ASoC: Intel: catpt: Wake up device before configuring SSP port ASoC: Intel: catpt: Relax clock selection conditions Codrin Ciubotariu (1): ASoC: mchp-spdiftx: Do not set Validity bit(s) Mark Brown (1): Merge tag 'v5.10-rc1' into asoc-5.10 Olivier Moysan (1): ASoC: cs42l51: manage mclk shutdown delay Srinivas Kandagatla (4): ASoC: codecs: wsa881x: add missing stream rates and format ASoC: qcom: sdm845: set driver name correctly ASoC: codecs: wcd934x: Set digital gain range correctly ASoC: codecs: wcd9335: Set digital gain range correctly Tomasz Figa (1): ASoC: Intel: kbl_rt5663_max98927: Fix kabylake_ssp_fixup function Tzung-Bi Shih (1): ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015 V Sujith Kumar Reddy (2): ASoC: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions ASoC: qcom: lpass-cpu: Fix clock disable failure sound/soc/atmel/mchp-spdiftx.c | 1 - sound/soc/codecs/cs42l51.c | 22 +++++++++++- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 ++ sound/soc/intel/Kconfig | 18 ---------- sound/soc/intel/atom/Makefile | 2 +- sound/soc/intel/atom/sst/Makefile | 6 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 +++++++++++++++++----- sound/soc/intel/catpt/dsp.c | 9 +++-- sound/soc/intel/catpt/pcm.c | 10 ++++++ sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 31 +++++++++++++---- sound/soc/qcom/lpass-cpu.c | 14 +++++--- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 2 ++ sound/soc/sof/loader.c | 5 +++ 16 files changed, 119 insertions(+), 48 deletions(-)

2 1

[PATCH] ASoC: q6afe-clocks: fix warning on symbol scope
by Srinivas Kandagatla 05 Nov '20

05 Nov '20

This patch fixes below warning when module is compiled with W=1 C=1 sound/soc/qcom/qdsp6/q6afe-clocks.c:122:18: warning: symbol 'q6afe_clks' was not declared. Should it be static? Fixes: 520a1c396d196 ("ASoC: q6afe-clocks: add q6afe clock controller") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index 2efc2eaa0424..87e4633afe2c 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -119,7 +119,7 @@ static const struct clk_ops clk_vote_q6afe_ops = { .unprepare = clk_unvote_q6afe_block, }; -struct q6afe_clk *q6afe_clks[Q6AFE_MAX_CLK_ID] = { +static struct q6afe_clk *q6afe_clks[Q6AFE_MAX_CLK_ID] = { [LPASS_CLK_ID_PRI_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_IBIT), [LPASS_CLK_ID_PRI_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_EBIT), [LPASS_CLK_ID_SEC_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_IBIT), -- 2.21.0

2 1

[PATCH] ASoc: qcom: lpass-cpu: fix warning on symbol scope
by Srinivas Kandagatla 05 Nov '20

05 Nov '20

This patch fixes below warning when module is compiled with W=1 C=1 lpass-cpu.c:677:22: warning: symbol 'lpass_hdmi_regmap_config' was not declared. Should it be static? Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 9d17c87445a9..1fd862e3fd8a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -674,7 +674,7 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) return false; } -struct regmap_config lpass_hdmi_regmap_config = { +static struct regmap_config lpass_hdmi_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, -- 2.21.0

2 1

[PATCH] ASoC: samsung: i2s: Remove redundant null check before clk_disable_unprepare
by Xu Wang 05 Nov '20

05 Nov '20

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/samsung/i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index df53d4ea808f..36969f0a3f9a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1212,8 +1212,7 @@ static int i2s_runtime_suspend(struct device *dev) priv->suspend_i2scon = readl(priv->addr + I2SCON); priv->suspend_i2spsr = readl(priv->addr + I2SPSR); - if (priv->op_clk) - clk_disable_unprepare(priv->op_clk); + clk_disable_unprepare(priv->op_clk); clk_disable_unprepare(priv->clk); return 0; -- 2.17.1

2 1

[PATCH] ASoC: pcm: DRAIN support reactivation
by Cezary Rojewski 05 Nov '20

05 Nov '20

soc-pcm's dpcm_fe_dai_do_trigger() supported DRAIN commnad up to kernel v5.4 where explicit switch(cmd) has been introduced which takes into account all SNDRV_PCM_TRIGGER_xxx but SNDRV_PCM_TRIGGER_DRAIN. Update switch statement to reactive support for it. As DRAIN is somewhat unique by lacking negative/stop counterpart, bring behaviour of dpcm_fe_dai_do_trigger() for said command back to its pre-v5.4 state by adding it to START/RESUME/PAUSE_RELEASE group. Fixes: acbf27746ecf ("ASoC: pcm: update FE/BE trigger order based on the command") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/soc-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..91bf33958159 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2231,6 +2231,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: @@ -2248,6 +2249,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; case SNDRV_PCM_TRIGGER_STOP: -- 2.17.1

3 2

[PATCH -next v2] ASoC: mediatek: mt8192: Make some symbols static
by Zou Wei 05 Nov '20

05 Nov '20

Fix the following sparse warnings: ./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static? ./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static? ./mt8192-afe-gpio.c:15:16: warning: symbol 'aud_pinctrl' was not declared. Should it be static? ./mt8192-afe-pcm.c:70:5: warning: symbol 'mt8192_get_memif_pbuf_size' was not declared. Should it be static? ./mt8192-afe-pcm.c:2137:39: warning: symbol 'mt8192_afe_component' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea00088..4208820 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -12,7 +12,7 @@ #include "mt8192-afe-common.h" #include "mt8192-afe-gpio.h" -struct pinctrl *aud_pinctrl; +static struct pinctrl *aud_pinctrl; enum mt8192_afe_gpio { MT8192_AFE_GPIO_DAT_MISO_OFF, diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 4a4729f..e7fec2d 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -67,7 +67,7 @@ static int mt8192_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) return mt8192_general_rate_transform(afe->dev, rate); } -int mt8192_get_memif_pbuf_size(struct snd_pcm_substream *substream) +static int mt8192_get_memif_pbuf_size(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -2134,7 +2134,7 @@ static int mt8192_afe_component_probe(struct snd_soc_component *component) return mtk_afe_add_sub_dai_control(component); } -const struct snd_soc_component_driver mt8192_afe_component = { +static const struct snd_soc_component_driver mt8192_afe_component = { .name = AFE_PCM_NAME, .probe = mt8192_afe_component_probe, .pointer = mtk_afe_pcm_pointer, diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c index 53c560e..5b29340 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c @@ -2037,7 +2037,7 @@ static const struct mtk_afe_i2s_priv mt8192_i2s_priv[DAI_I2S_NUM] = { }, }; -int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) { struct mt8192_afe_private *afe_priv = afe->platform_priv; const struct device_node *of_node = afe->dev->of_node; @@ -2057,7 +2057,7 @@ int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) return 0; } -int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) { int i; int ret; -- 2.6.2

3 2

[PATCH] ASoC: mediatek: mt8192: Fix build failure
by Sudip Mukherjee 05 Nov '20

05 Nov '20

A build of arm64 allmodconfig with next-20201105 fails with the error: ERROR: modpost: "mt8192_afe_gpio_request" undefined! ERROR: modpost: "mt8192_afe_gpio_init" undefined! Export the symbols so that mt8192-mt6359-rt1015-rt5682.ko finds it. Signed-off-by: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> --- build log at: https://travis-ci.com/github/sudipm-mukherjee/linux-test/jobs/428486008 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea000888c9e8..fbbe9ed9adb3 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -160,6 +160,7 @@ int mt8192_afe_gpio_init(struct device *dev) return 0; } +EXPORT_SYMBOL(mt8192_afe_gpio_init); static int mt8192_afe_gpio_adda_dl(struct device *dev, bool enable) { @@ -304,3 +305,4 @@ int mt8192_afe_gpio_request(struct device *dev, bool enable, return 0; } +EXPORT_SYMBOL(mt8192_afe_gpio_request); -- 2.11.0

2 1

[PATCH] ASoC: pxa: pxa-ssp: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare
by Xu Wang 05 Nov '20

05 Nov '20

ecause clk_prepare_enable() and clk_disable_unprepare() already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/pxa/pxa-ssp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c4e7307a4437..b941adcbb8f9 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -99,8 +99,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - if (priv->extclk) - clk_prepare_enable(priv->extclk); + clk_prepare_enable(priv->extclk); dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) @@ -124,8 +123,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } - if (priv->extclk) - clk_disable_unprepare(priv->extclk); + clk_disable_unprepare(priv->extclk); kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); -- 2.17.1

2 1

[PATCH -next] ASoC: mediatek: mt8192: Make some symbols static
by Zou Wei 05 Nov '20

05 Nov '20

Fix the following sparse warnings: ./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static? ./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c index 53c560e..5b29340 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c @@ -2037,7 +2037,7 @@ static const struct mtk_afe_i2s_priv mt8192_i2s_priv[DAI_I2S_NUM] = { }, }; -int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) { struct mt8192_afe_private *afe_priv = afe->platform_priv; const struct device_node *of_node = afe->dev->of_node; @@ -2057,7 +2057,7 @@ int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) return 0; } -int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) { int i; int ret; -- 2.6.2

1 0

[PATCH] sound: soc: mediatek: mt8192: fix modpost ERROR
by Anders Roxell 05 Nov '20

05 Nov '20

When enabling CONFIG_SND_SOC_MT8192_MT6359_RT1015_RT5682=m the following error shows up: ERROR: modpost: "mt8192_afe_gpio_request" [sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.ko] undefined! ERROR: modpost: "mt8192_afe_gpio_init" [sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.ko] undefined! Fix the link error by export the symbols mt8192_afe_gpio_init and mt8192_afe_gpio_request. Fixes: 18b13ff23fab ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea000888c9e8..0286c95ed2b8 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -160,6 +160,7 @@ int mt8192_afe_gpio_init(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(mt8192_afe_gpio_init); static int mt8192_afe_gpio_adda_dl(struct device *dev, bool enable) { @@ -304,3 +305,4 @@ int mt8192_afe_gpio_request(struct device *dev, bool enable, return 0; } +EXPORT_SYMBOL_GPL(mt8192_afe_gpio_request); -- 2.28.0

1 0

[PATCH 0/8] slab: provide and use krealloc_array()
by Bartosz Golaszewski 05 Nov '20

05 Nov '20

From: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy brought to my attention the fact that users allocating an array of equally sized elements should check if the size multiplication doesn't overflow. This is why we have helpers like kmalloc_array(). However we don't have krealloc_array() equivalent and there are many users who do their own multiplication when calling krealloc() for arrays. This series provides krealloc_array() and uses it in a couple places. A separate series will follow adding devm_krealloc_array() which is needed in the xilinx adc driver. Bartosz Golaszewski (8): mm: slab: provide krealloc_array() ALSA: pcm: use krealloc_array() vhost: vringh: use krealloc_array() pinctrl: use krealloc_array() edac: ghes: use krealloc_array() drm: atomic: use krealloc_array() hwtracing: intel: use krealloc_array() dma-buf: use krealloc_array() drivers/dma-buf/sync_file.c | 4 ++-- drivers/edac/ghes_edac.c | 4 ++-- drivers/gpu/drm/drm_atomic.c | 3 ++- drivers/hwtracing/intel_th/msu.c | 2 +- drivers/pinctrl/pinctrl-utils.c | 2 +- drivers/vhost/vringh.c | 3 ++- include/linux/slab.h | 11 +++++++++++ sound/core/pcm_lib.c | 4 ++-- 8 files changed, 23 insertions(+), 10 deletions(-) -- 2.29.1

11 20

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for Qu-16
by Geoffrey D. Bennett 05 Nov '20

05 Nov '20

This patch fixes audio distortion on playback for the Allen&Heath Qu-16. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee8..c4915de 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -336,6 +336,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */ ep = 0x81; ifnum = 3; goto add_sync_ep_from_ifnum;

2 1

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for MODX
by Geoffrey D. Bennett 05 Nov '20

05 Nov '20

This patch fixes audio distortion on playback for the Yamaha MODX. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Tested-by: Frank Slotta <frank.slotta(a)posteo.de> --- pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..22fbd1f48b3e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -345,6 +345,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum;

2 3

[PATCH v2 0/6] ASoC: codecs: add support for LPASS Codec macros
by Srinivas Kandagatla 04 Nov '20

04 Nov '20

This patchset adds support for two Codec Macro blocks( WSA and VA) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD Codecs to provide headphone/ear/lineout etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Thanks, srini -Changes since v1: - updated code to fix cppcheck errors suggested by Pierre - fixed various coding style and variable initialization suggested by Pierre. - yaml bindings clock names replaced with constants. Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes .../bindings/sound/qcom,lpass-va-macro.yaml | 67 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 69 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-va-macro.c | 1500 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2472 +++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 15 + 7 files changed, 4135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h -- 2.21.0

3 15

[PATCH v5 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 04 Nov '20

04 Nov '20

Hi Rob, Mark Cc: Sameer These are v5 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v4 -> v5 - move "endpoint" under "port" v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 162 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 620 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

3 7

[PATCH v3] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 04 Nov '20

04 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 ++++++++++++++++ 2 files changed, 143 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..a9cd6ad0bf5a 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,106 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 2: + val |= RT1015_I2S_TX_2CH; + break; + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 16: + val |= RT1015_I2S_CH_TX_LEN_16B; + break; + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + /* This is an assumption that the system sends stereo audio to the amplifier typically. + * And the stereo audio is placed in slot 0/2/4/6 as the starting slot. + * The users could select the channel from L/R/L+R by "Mono LR Select" control. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1075,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

3 2

[PATCH v3 0/7] ASoC: topology: Change to resource managed memory
by Amadeusz Sławiński 04 Nov '20

04 Nov '20

Almost all other allocations in ASoC API are resource managed, the only exception is soc-topology.c. This patchset clean ups few unnecessary functions in preparation for change and then changes to devm_ functions for allocation. Amadeusz Sławiński (6): ASoC: topology: Remove unused functions from topology API ASoC: topology: Remove multistep topology loading ASoC: topology: Unify all device references ASoC: topology: Change allocations to resource managed ASoC: topology: Remove empty functions ASoC: topology: Simplify remove_widget function include/sound/soc-topology.h | 13 +- sound/soc/intel/skylake/skl-topology.c | 9 +- sound/soc/soc-topology.c | 302 +++++-------------------- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/topology.c | 4 +- 5 files changed, 57 insertions(+), 273 deletions(-) -- 2.25.1

3 10

[PATCH] ASoC: mchp-spdiftx: Do not set Validity bit(s)
by Codrin Ciubotariu 04 Nov '20

04 Nov '20

The Validity bits (bit 28) must not be set in order to have the samples valid. Some controllers look for this bit and ignore the samples if it is set. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/mchp-spdiftx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 82c1eecd2528..3bd350afb743 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream, } mchp_spdiftx_channel_status_write(dev); spin_unlock_irqrestore(&ctrl->lock, flags); - mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2; if (dev->gclk_enabled) { clk_disable_unprepare(dev->gclk); -- 2.27.0

2 1

[Feature Request] Add option to change Audio Mixer Source node (0x0c, 0x0d, 0x0e etc)
by GitHub issues - opened 04 Nov '20

04 Nov '20

alsa-project/alsa-tools issue #3 was opened from kevinsmia1939: Hello, I am looking for linux audio tool that can change Audio Mixer Source node, 0x0c to 0x0d for example. HDA Analyzer can do this (right side in red circle of screenshot) https://i.imgur.com/UcELkjU.png But HDA analyzer use python 2 and some other depreciate python modules and since I reinstall my machine (openSUSE TW), I lost the old packages (install old packages manually does not work for some reason). https://www.alsa-project.org/main/index.php/HDA_Analyzer I tried HDAJackRetask, but could not find the option or feature to do it. However, HDAJackRetask is the closest thing to HDA Analyzer, so I think it would make sense to ask for this feature here. If there are already tools that can do this (other than HDA Analyzer) please tell me. Thank you. Issue URL : https://github.com/alsa-project/alsa-tools/issues/3 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Alternative to HDA Analyzer that can change Audio Mixer Source Node?
by andythe_great 04 Nov '20

04 Nov '20

Hello, I need a tool that can change Audio Mixer Source node as seen in this screenshot circle in red.

1 0

[PATCH] sound/core/seq: remove useless function
by Yu Hao 04 Nov '20

04 Nov '20

The function snd_seq_queue_client_termination() is only called from the function seq_free_client1(). The function seq_free_client1() calls the function snd_seq_queue_client_leave() and the function snd_seq_queue_client_termination() together. Because the function snd_seq_queue_client_leave() does all things, so the function snd_seq_queue_client_termination() is a useless function. Signed-off-by: Yu Hao <yuhaobehappy(a)gmail.com> --- sound/core/seq/seq_clientmgr.c | 1 - sound/core/seq/seq_queue.c | 27 --------------------------- sound/core/seq/seq_queue.h | 3 --- 3 files changed, 31 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index cc93157fa950..f9f2fea58b32 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -279,7 +279,6 @@ static int seq_free_client1(struct snd_seq_client *client) snd_seq_delete_all_ports(client); snd_seq_queue_client_leave(client->number); snd_use_lock_sync(&client->use_lock); - snd_seq_queue_client_termination(client->number); if (client->pool) snd_seq_pool_delete(&client->pool); spin_lock_irq(&clients_lock); diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 71a6ea62c3be..13cfc2d47fa7 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -537,33 +537,6 @@ int snd_seq_queue_is_used(int queueid, int client) /*----------------------------------------------------------------*/ -/* notification that client has left the system - - * stop the timer on all queues owned by this client - */ -void snd_seq_queue_client_termination(int client) -{ - unsigned long flags; - int i; - struct snd_seq_queue *q; - bool matched; - - for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { - if ((q = queueptr(i)) == NULL) - continue; - spin_lock_irqsave(&q->owner_lock, flags); - matched = (q->owner == client); - if (matched) - q->klocked = 1; - spin_unlock_irqrestore(&q->owner_lock, flags); - if (matched) { - if (q->timer->running) - snd_seq_timer_stop(q->timer); - snd_seq_timer_reset(q->timer); - } - queuefree(q); - } -} - /* final stage notification - * remove cells for no longer exist client (for non-owned queue) * or delete this queue (for owned queue) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 9254c8dbe5e3..1c3a8d3254d9 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -59,9 +59,6 @@ struct snd_seq_queue *snd_seq_queue_alloc(int client, int locked, unsigned int f /* delete queue (destructor) */ int snd_seq_queue_delete(int client, int queueid); -/* notification that client has left the system */ -void snd_seq_queue_client_termination(int client); - /* final stage */ void snd_seq_queue_client_leave(int client); -- 2.17.1

2 2

[PATCH v2] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 04 Nov '20

04 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 ++++++++++++++ sound/soc/codecs/rt1015.c | 20 +++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 4 files changed, 43 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt index fcfd02d8d32f..da9697c80d2b 100644 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -8,10 +8,16 @@ Required properties: - reg : The I2C address of the device. +Optional properties: + +- realtek,realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. Example: rt1015: codec@28 { compatible = "realtek,rt1015"; reg = <0x28>; + realtek,power-up-delay-ms = <50>; }; diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..70a7538d4c89 --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay_ms; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..967193518349 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay_ms = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay_ms); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay-ms", + &rt1015->pdata.power_up_delay_ms); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

2 1

[PATCH] mfd: madera: Delete register field xxx_WIDTH defines
by Richard Fitzgerald 04 Nov '20

04 Nov '20

The register field xxx_WIDTH defined are not used in current code. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/linux/mfd/madera/registers.h | 635 --------------------------- 1 file changed, 635 deletions(-) diff --git a/include/linux/mfd/madera/registers.h b/include/linux/mfd/madera/registers.h index fe909d177762..b44aeb461d0c 100644 --- a/include/linux/mfd/madera/registers.h +++ b/include/linux/mfd/madera/registers.h @@ -1286,566 +1286,438 @@ /* (0x0000) Software_Reset */ #define MADERA_SW_RST_DEV_ID1_MASK 0xFFFF #define MADERA_SW_RST_DEV_ID1_SHIFT 0 -#define MADERA_SW_RST_DEV_ID1_WIDTH 16 /* (0x0001) Hardware_Revision */ #define MADERA_HW_REVISION_MASK 0x00FF #define MADERA_HW_REVISION_SHIFT 0 -#define MADERA_HW_REVISION_WIDTH 8 /* (0x0020) Tone_Generator_1 */ #define MADERA_TONE2_ENA 0x0002 #define MADERA_TONE2_ENA_MASK 0x0002 #define MADERA_TONE2_ENA_SHIFT 1 -#define MADERA_TONE2_ENA_WIDTH 1 #define MADERA_TONE1_ENA 0x0001 #define MADERA_TONE1_ENA_MASK 0x0001 #define MADERA_TONE1_ENA_SHIFT 0 -#define MADERA_TONE1_ENA_WIDTH 1 /* (0x0021) Tone_Generator_2 */ #define MADERA_TONE1_LVL_0_MASK 0xFFFF #define MADERA_TONE1_LVL_0_SHIFT 0 -#define MADERA_TONE1_LVL_0_WIDTH 16 /* (0x0022) Tone_Generator_3 */ #define MADERA_TONE1_LVL_MASK 0x00FF #define MADERA_TONE1_LVL_SHIFT 0 -#define MADERA_TONE1_LVL_WIDTH 8 /* (0x0023) Tone_Generator_4 */ #define MADERA_TONE2_LVL_0_MASK 0xFFFF #define MADERA_TONE2_LVL_0_SHIFT 0 -#define MADERA_TONE2_LVL_0_WIDTH 16 /* (0x0024) Tone_Generator_5 */ #define MADERA_TONE2_LVL_MASK 0x00FF #define MADERA_TONE2_LVL_SHIFT 0 -#define MADERA_TONE2_LVL_WIDTH 8 /* (0x0030) PWM_Drive_1 */ #define MADERA_PWM2_ENA 0x0002 #define MADERA_PWM2_ENA_MASK 0x0002 #define MADERA_PWM2_ENA_SHIFT 1 -#define MADERA_PWM2_ENA_WIDTH 1 #define MADERA_PWM1_ENA 0x0001 #define MADERA_PWM1_ENA_MASK 0x0001 #define MADERA_PWM1_ENA_SHIFT 0 -#define MADERA_PWM1_ENA_WIDTH 1 /* (0x00A0) Comfort_Noise_Generator */ #define MADERA_NOISE_GEN_ENA 0x0020 #define MADERA_NOISE_GEN_ENA_MASK 0x0020 #define MADERA_NOISE_GEN_ENA_SHIFT 5 -#define MADERA_NOISE_GEN_ENA_WIDTH 1 #define MADERA_NOISE_GEN_GAIN_MASK 0x001F #define MADERA_NOISE_GEN_GAIN_SHIFT 0 -#define MADERA_NOISE_GEN_GAIN_WIDTH 5 /* (0x0100) Clock_32k_1 */ #define MADERA_CLK_32K_ENA 0x0040 #define MADERA_CLK_32K_ENA_MASK 0x0040 #define MADERA_CLK_32K_ENA_SHIFT 6 -#define MADERA_CLK_32K_ENA_WIDTH 1 #define MADERA_CLK_32K_SRC_MASK 0x0003 #define MADERA_CLK_32K_SRC_SHIFT 0 -#define MADERA_CLK_32K_SRC_WIDTH 2 /* (0x0101) System_Clock_1 */ #define MADERA_SYSCLK_FRAC 0x8000 #define MADERA_SYSCLK_FRAC_MASK 0x8000 #define MADERA_SYSCLK_FRAC_SHIFT 15 -#define MADERA_SYSCLK_FRAC_WIDTH 1 #define MADERA_SYSCLK_FREQ_MASK 0x0700 #define MADERA_SYSCLK_FREQ_SHIFT 8 -#define MADERA_SYSCLK_FREQ_WIDTH 3 #define MADERA_SYSCLK_ENA 0x0040 #define MADERA_SYSCLK_ENA_MASK 0x0040 #define MADERA_SYSCLK_ENA_SHIFT 6 -#define MADERA_SYSCLK_ENA_WIDTH 1 #define MADERA_SYSCLK_SRC_MASK 0x000F #define MADERA_SYSCLK_SRC_SHIFT 0 -#define MADERA_SYSCLK_SRC_WIDTH 4 /* (0x0102) Sample_rate_1 */ #define MADERA_SAMPLE_RATE_1_MASK 0x001F #define MADERA_SAMPLE_RATE_1_SHIFT 0 -#define MADERA_SAMPLE_RATE_1_WIDTH 5 /* (0x0103) Sample_rate_2 */ #define MADERA_SAMPLE_RATE_2_MASK 0x001F #define MADERA_SAMPLE_RATE_2_SHIFT 0 -#define MADERA_SAMPLE_RATE_2_WIDTH 5 /* (0x0104) Sample_rate_3 */ #define MADERA_SAMPLE_RATE_3_MASK 0x001F #define MADERA_SAMPLE_RATE_3_SHIFT 0 -#define MADERA_SAMPLE_RATE_3_WIDTH 5 /* (0x0112) Async_clock_1 */ #define MADERA_ASYNC_CLK_FREQ_MASK 0x0700 #define MADERA_ASYNC_CLK_FREQ_SHIFT 8 -#define MADERA_ASYNC_CLK_FREQ_WIDTH 3 #define MADERA_ASYNC_CLK_ENA 0x0040 #define MADERA_ASYNC_CLK_ENA_MASK 0x0040 #define MADERA_ASYNC_CLK_ENA_SHIFT 6 -#define MADERA_ASYNC_CLK_ENA_WIDTH 1 #define MADERA_ASYNC_CLK_SRC_MASK 0x000F #define MADERA_ASYNC_CLK_SRC_SHIFT 0 -#define MADERA_ASYNC_CLK_SRC_WIDTH 4 /* (0x0113) Async_sample_rate_1 */ #define MADERA_ASYNC_SAMPLE_RATE_1_MASK 0x001F #define MADERA_ASYNC_SAMPLE_RATE_1_SHIFT 0 -#define MADERA_ASYNC_SAMPLE_RATE_1_WIDTH 5 /* (0x0114) Async_sample_rate_2 */ #define MADERA_ASYNC_SAMPLE_RATE_2_MASK 0x001F #define MADERA_ASYNC_SAMPLE_RATE_2_SHIFT 0 -#define MADERA_ASYNC_SAMPLE_RATE_2_WIDTH 5 /* (0x0120) DSP_Clock_1 */ #define MADERA_DSP_CLK_FREQ_LEGACY 0x0700 #define MADERA_DSP_CLK_FREQ_LEGACY_MASK 0x0700 #define MADERA_DSP_CLK_FREQ_LEGACY_SHIFT 8 -#define MADERA_DSP_CLK_FREQ_LEGACY_WIDTH 3 #define MADERA_DSP_CLK_ENA 0x0040 #define MADERA_DSP_CLK_ENA_MASK 0x0040 #define MADERA_DSP_CLK_ENA_SHIFT 6 -#define MADERA_DSP_CLK_ENA_WIDTH 1 #define MADERA_DSP_CLK_SRC 0x000F #define MADERA_DSP_CLK_SRC_MASK 0x000F #define MADERA_DSP_CLK_SRC_SHIFT 0 -#define MADERA_DSP_CLK_SRC_WIDTH 4 /* (0x0122) DSP_Clock_2 */ #define MADERA_DSP_CLK_FREQ_MASK 0x03FF #define MADERA_DSP_CLK_FREQ_SHIFT 0 -#define MADERA_DSP_CLK_FREQ_WIDTH 10 /* (0x0149) Output_system_clock */ #define MADERA_OPCLK_ENA 0x8000 #define MADERA_OPCLK_ENA_MASK 0x8000 #define MADERA_OPCLK_ENA_SHIFT 15 -#define MADERA_OPCLK_ENA_WIDTH 1 #define MADERA_OPCLK_DIV_MASK 0x00F8 #define MADERA_OPCLK_DIV_SHIFT 3 -#define MADERA_OPCLK_DIV_WIDTH 5 #define MADERA_OPCLK_SEL_MASK 0x0007 #define MADERA_OPCLK_SEL_SHIFT 0 -#define MADERA_OPCLK_SEL_WIDTH 3 /* (0x014A) Output_async_clock */ #define MADERA_OPCLK_ASYNC_ENA 0x8000 #define MADERA_OPCLK_ASYNC_ENA_MASK 0x8000 #define MADERA_OPCLK_ASYNC_ENA_SHIFT 15 -#define MADERA_OPCLK_ASYNC_ENA_WIDTH 1 #define MADERA_OPCLK_ASYNC_DIV_MASK 0x00F8 #define MADERA_OPCLK_ASYNC_DIV_SHIFT 3 -#define MADERA_OPCLK_ASYNC_DIV_WIDTH 5 #define MADERA_OPCLK_ASYNC_SEL_MASK 0x0007 #define MADERA_OPCLK_ASYNC_SEL_SHIFT 0 -#define MADERA_OPCLK_ASYNC_SEL_WIDTH 3 /* (0x0171) FLL1_Control_1 */ #define CS47L92_FLL1_REFCLK_SRC_MASK 0xF000 #define CS47L92_FLL1_REFCLK_SRC_SHIFT 12 -#define CS47L92_FLL1_REFCLK_SRC_WIDTH 4 #define MADERA_FLL1_HOLD_MASK 0x0004 #define MADERA_FLL1_HOLD_SHIFT 2 -#define MADERA_FLL1_HOLD_WIDTH 1 #define MADERA_FLL1_FREERUN 0x0002 #define MADERA_FLL1_FREERUN_MASK 0x0002 #define MADERA_FLL1_FREERUN_SHIFT 1 -#define MADERA_FLL1_FREERUN_WIDTH 1 #define MADERA_FLL1_ENA 0x0001 #define MADERA_FLL1_ENA_MASK 0x0001 #define MADERA_FLL1_ENA_SHIFT 0 -#define MADERA_FLL1_ENA_WIDTH 1 /* (0x0172) FLL1_Control_2 */ #define MADERA_FLL1_CTRL_UPD 0x8000 #define MADERA_FLL1_CTRL_UPD_MASK 0x8000 #define MADERA_FLL1_CTRL_UPD_SHIFT 15 -#define MADERA_FLL1_CTRL_UPD_WIDTH 1 #define MADERA_FLL1_N_MASK 0x03FF #define MADERA_FLL1_N_SHIFT 0 -#define MADERA_FLL1_N_WIDTH 10 /* (0x0173) FLL1_Control_3 */ #define MADERA_FLL1_THETA_MASK 0xFFFF #define MADERA_FLL1_THETA_SHIFT 0 -#define MADERA_FLL1_THETA_WIDTH 16 /* (0x0174) FLL1_Control_4 */ #define MADERA_FLL1_LAMBDA_MASK 0xFFFF #define MADERA_FLL1_LAMBDA_SHIFT 0 -#define MADERA_FLL1_LAMBDA_WIDTH 16 /* (0x0175) FLL1_Control_5 */ #define MADERA_FLL1_FRATIO_MASK 0x0F00 #define MADERA_FLL1_FRATIO_SHIFT 8 -#define MADERA_FLL1_FRATIO_WIDTH 4 #define MADERA_FLL1_FB_DIV_MASK 0x03FF #define MADERA_FLL1_FB_DIV_SHIFT 0 -#define MADERA_FLL1_FB_DIV_WIDTH 10 /* (0x0176) FLL1_Control_6 */ #define MADERA_FLL1_REFCLK_DIV_MASK 0x00C0 #define MADERA_FLL1_REFCLK_DIV_SHIFT 6 -#define MADERA_FLL1_REFCLK_DIV_WIDTH 2 #define MADERA_FLL1_REFCLK_SRC_MASK 0x000F #define MADERA_FLL1_REFCLK_SRC_SHIFT 0 -#define MADERA_FLL1_REFCLK_SRC_WIDTH 4 /* (0x0179) FLL1_Control_7 */ #define MADERA_FLL1_GAIN_MASK 0x003c #define MADERA_FLL1_GAIN_SHIFT 2 -#define MADERA_FLL1_GAIN_WIDTH 4 /* (0x017A) FLL1_EFS_2 */ #define MADERA_FLL1_PHASE_GAIN_MASK 0xF000 #define MADERA_FLL1_PHASE_GAIN_SHIFT 12 -#define MADERA_FLL1_PHASE_GAIN_WIDTH 4 #define MADERA_FLL1_PHASE_ENA_MASK 0x0800 #define MADERA_FLL1_PHASE_ENA_SHIFT 11 -#define MADERA_FLL1_PHASE_ENA_WIDTH 1 /* (0x017A) FLL1_Control_10 */ #define MADERA_FLL1_HP_MASK 0xC000 #define MADERA_FLL1_HP_SHIFT 14 -#define MADERA_FLL1_HP_WIDTH 2 #define MADERA_FLL1_PHASEDET_ENA_MASK 0x1000 #define MADERA_FLL1_PHASEDET_ENA_SHIFT 12 -#define MADERA_FLL1_PHASEDET_ENA_WIDTH 1 /* (0x017B) FLL1_Control_11 */ #define MADERA_FLL1_LOCKDET_THR_MASK 0x001E #define MADERA_FLL1_LOCKDET_THR_SHIFT 1 -#define MADERA_FLL1_LOCKDET_THR_WIDTH 4 #define MADERA_FLL1_LOCKDET_MASK 0x0001 #define MADERA_FLL1_LOCKDET_SHIFT 0 -#define MADERA_FLL1_LOCKDET_WIDTH 1 /* (0x017D) FLL1_Digital_Test_1 */ #define MADERA_FLL1_SYNC_EFS_ENA_MASK 0x0100 #define MADERA_FLL1_SYNC_EFS_ENA_SHIFT 8 -#define MADERA_FLL1_SYNC_EFS_ENA_WIDTH 1 #define MADERA_FLL1_CLK_VCO_FAST_SRC_MASK 0x0003 #define MADERA_FLL1_CLK_VCO_FAST_SRC_SHIFT 0 -#define MADERA_FLL1_CLK_VCO_FAST_SRC_WIDTH 2 /* (0x0181) FLL1_Synchroniser_1 */ #define MADERA_FLL1_SYNC_ENA 0x0001 #define MADERA_FLL1_SYNC_ENA_MASK 0x0001 #define MADERA_FLL1_SYNC_ENA_SHIFT 0 -#define MADERA_FLL1_SYNC_ENA_WIDTH 1 /* (0x0182) FLL1_Synchroniser_2 */ #define MADERA_FLL1_SYNC_N_MASK 0x03FF #define MADERA_FLL1_SYNC_N_SHIFT 0 -#define MADERA_FLL1_SYNC_N_WIDTH 10 /* (0x0183) FLL1_Synchroniser_3 */ #define MADERA_FLL1_SYNC_THETA_MASK 0xFFFF #define MADERA_FLL1_SYNC_THETA_SHIFT 0 -#define MADERA_FLL1_SYNC_THETA_WIDTH 16 /* (0x0184) FLL1_Synchroniser_4 */ #define MADERA_FLL1_SYNC_LAMBDA_MASK 0xFFFF #define MADERA_FLL1_SYNC_LAMBDA_SHIFT 0 -#define MADERA_FLL1_SYNC_LAMBDA_WIDTH 16 /* (0x0185) FLL1_Synchroniser_5 */ #define MADERA_FLL1_SYNC_FRATIO_MASK 0x0700 #define MADERA_FLL1_SYNC_FRATIO_SHIFT 8 -#define MADERA_FLL1_SYNC_FRATIO_WIDTH 3 /* (0x0186) FLL1_Synchroniser_6 */ #define MADERA_FLL1_SYNCCLK_DIV_MASK 0x00C0 #define MADERA_FLL1_SYNCCLK_DIV_SHIFT 6 -#define MADERA_FLL1_SYNCCLK_DIV_WIDTH 2 #define MADERA_FLL1_SYNCCLK_SRC_MASK 0x000F #define MADERA_FLL1_SYNCCLK_SRC_SHIFT 0 -#define MADERA_FLL1_SYNCCLK_SRC_WIDTH 4 /* (0x0187) FLL1_Synchroniser_7 */ #define MADERA_FLL1_SYNC_GAIN_MASK 0x003c #define MADERA_FLL1_SYNC_GAIN_SHIFT 2 -#define MADERA_FLL1_SYNC_GAIN_WIDTH 4 #define MADERA_FLL1_SYNC_DFSAT 0x0001 #define MADERA_FLL1_SYNC_DFSAT_MASK 0x0001 #define MADERA_FLL1_SYNC_DFSAT_SHIFT 0 -#define MADERA_FLL1_SYNC_DFSAT_WIDTH 1 /* (0x01D1) FLL_AO_Control_1 */ #define MADERA_FLL_AO_HOLD 0x0004 #define MADERA_FLL_AO_HOLD_MASK 0x0004 #define MADERA_FLL_AO_HOLD_SHIFT 2 -#define MADERA_FLL_AO_HOLD_WIDTH 1 #define MADERA_FLL_AO_FREERUN 0x0002 #define MADERA_FLL_AO_FREERUN_MASK 0x0002 #define MADERA_FLL_AO_FREERUN_SHIFT 1 -#define MADERA_FLL_AO_FREERUN_WIDTH 1 #define MADERA_FLL_AO_ENA 0x0001 #define MADERA_FLL_AO_ENA_MASK 0x0001 #define MADERA_FLL_AO_ENA_SHIFT 0 -#define MADERA_FLL_AO_ENA_WIDTH 1 /* (0x01D2) FLL_AO_Control_2 */ #define MADERA_FLL_AO_CTRL_UPD 0x8000 #define MADERA_FLL_AO_CTRL_UPD_MASK 0x8000 #define MADERA_FLL_AO_CTRL_UPD_SHIFT 15 -#define MADERA_FLL_AO_CTRL_UPD_WIDTH 1 /* (0x01D6) FLL_AO_Control_6 */ #define MADERA_FLL_AO_REFCLK_SRC_MASK 0x000F #define MADERA_FLL_AO_REFCLK_SRC_SHIFT 0 -#define MADERA_FLL_AO_REFCLK_SRC_WIDTH 4 /* (0x0200) Mic_Charge_Pump_1 */ #define MADERA_CPMIC_BYPASS 0x0002 #define MADERA_CPMIC_BYPASS_MASK 0x0002 #define MADERA_CPMIC_BYPASS_SHIFT 1 -#define MADERA_CPMIC_BYPASS_WIDTH 1 #define MADERA_CPMIC_ENA 0x0001 #define MADERA_CPMIC_ENA_MASK 0x0001 #define MADERA_CPMIC_ENA_SHIFT 0 -#define MADERA_CPMIC_ENA_WIDTH 1 /* (0x0210) LDO1_Control_1 */ #define MADERA_LDO1_VSEL_MASK 0x07E0 #define MADERA_LDO1_VSEL_SHIFT 5 -#define MADERA_LDO1_VSEL_WIDTH 6 #define MADERA_LDO1_FAST 0x0010 #define MADERA_LDO1_FAST_MASK 0x0010 #define MADERA_LDO1_FAST_SHIFT 4 -#define MADERA_LDO1_FAST_WIDTH 1 #define MADERA_LDO1_DISCH 0x0004 #define MADERA_LDO1_DISCH_MASK 0x0004 #define MADERA_LDO1_DISCH_SHIFT 2 -#define MADERA_LDO1_DISCH_WIDTH 1 #define MADERA_LDO1_BYPASS 0x0002 #define MADERA_LDO1_BYPASS_MASK 0x0002 #define MADERA_LDO1_BYPASS_SHIFT 1 -#define MADERA_LDO1_BYPASS_WIDTH 1 #define MADERA_LDO1_ENA 0x0001 #define MADERA_LDO1_ENA_MASK 0x0001 #define MADERA_LDO1_ENA_SHIFT 0 -#define MADERA_LDO1_ENA_WIDTH 1 /* (0x0213) LDO2_Control_1 */ #define MADERA_LDO2_VSEL_MASK 0x07E0 #define MADERA_LDO2_VSEL_SHIFT 5 -#define MADERA_LDO2_VSEL_WIDTH 6 #define MADERA_LDO2_FAST 0x0010 #define MADERA_LDO2_FAST_MASK 0x0010 #define MADERA_LDO2_FAST_SHIFT 4 -#define MADERA_LDO2_FAST_WIDTH 1 #define MADERA_LDO2_DISCH 0x0004 #define MADERA_LDO2_DISCH_MASK 0x0004 #define MADERA_LDO2_DISCH_SHIFT 2 -#define MADERA_LDO2_DISCH_WIDTH 1 #define MADERA_LDO2_BYPASS 0x0002 #define MADERA_LDO2_BYPASS_MASK 0x0002 #define MADERA_LDO2_BYPASS_SHIFT 1 -#define MADERA_LDO2_BYPASS_WIDTH 1 #define MADERA_LDO2_ENA 0x0001 #define MADERA_LDO2_ENA_MASK 0x0001 #define MADERA_LDO2_ENA_SHIFT 0 -#define MADERA_LDO2_ENA_WIDTH 1 /* (0x0218) Mic_Bias_Ctrl_1 */ #define MADERA_MICB1_EXT_CAP 0x8000 #define MADERA_MICB1_EXT_CAP_MASK 0x8000 #define MADERA_MICB1_EXT_CAP_SHIFT 15 -#define MADERA_MICB1_EXT_CAP_WIDTH 1 #define MADERA_MICB1_LVL_MASK 0x01E0 #define MADERA_MICB1_LVL_SHIFT 5 -#define MADERA_MICB1_LVL_WIDTH 4 #define MADERA_MICB1_ENA 0x0001 #define MADERA_MICB1_ENA_MASK 0x0001 #define MADERA_MICB1_ENA_SHIFT 0 -#define MADERA_MICB1_ENA_WIDTH 1 /* (0x021C) Mic_Bias_Ctrl_5 */ #define MADERA_MICB1D_ENA 0x1000 #define MADERA_MICB1D_ENA_MASK 0x1000 #define MADERA_MICB1D_ENA_SHIFT 12 -#define MADERA_MICB1D_ENA_WIDTH 1 #define MADERA_MICB1C_ENA 0x0100 #define MADERA_MICB1C_ENA_MASK 0x0100 #define MADERA_MICB1C_ENA_SHIFT 8 -#define MADERA_MICB1C_ENA_WIDTH 1 #define MADERA_MICB1B_ENA 0x0010 #define MADERA_MICB1B_ENA_MASK 0x0010 #define MADERA_MICB1B_ENA_SHIFT 4 -#define MADERA_MICB1B_ENA_WIDTH 1 #define MADERA_MICB1A_ENA 0x0001 #define MADERA_MICB1A_ENA_MASK 0x0001 #define MADERA_MICB1A_ENA_SHIFT 0 -#define MADERA_MICB1A_ENA_WIDTH 1 /* (0x021E) Mic_Bias_Ctrl_6 */ #define MADERA_MICB2D_ENA 0x1000 #define MADERA_MICB2D_ENA_MASK 0x1000 #define MADERA_MICB2D_ENA_SHIFT 12 -#define MADERA_MICB2D_ENA_WIDTH 1 #define MADERA_MICB2C_ENA 0x0100 #define MADERA_MICB2C_ENA_MASK 0x0100 #define MADERA_MICB2C_ENA_SHIFT 8 -#define MADERA_MICB2C_ENA_WIDTH 1 #define MADERA_MICB2B_ENA 0x0010 #define MADERA_MICB2B_ENA_MASK 0x0010 #define MADERA_MICB2B_ENA_SHIFT 4 -#define MADERA_MICB2B_ENA_WIDTH 1 #define MADERA_MICB2A_ENA 0x0001 #define MADERA_MICB2A_ENA_MASK 0x0001 #define MADERA_MICB2A_ENA_SHIFT 0 -#define MADERA_MICB2A_ENA_WIDTH 1 /* (0x0225) - HP Ctrl 1L */ #define MADERA_RMV_SHRT_HP1L 0x4000 #define MADERA_RMV_SHRT_HP1L_MASK 0x4000 #define MADERA_RMV_SHRT_HP1L_SHIFT 14 -#define MADERA_RMV_SHRT_HP1L_WIDTH 1 #define MADERA_HP1L_FLWR 0x0004 #define MADERA_HP1L_FLWR_MASK 0x0004 #define MADERA_HP1L_FLWR_SHIFT 2 -#define MADERA_HP1L_FLWR_WIDTH 1 #define MADERA_HP1L_SHRTI 0x0002 #define MADERA_HP1L_SHRTI_MASK 0x0002 #define MADERA_HP1L_SHRTI_SHIFT 1 -#define MADERA_HP1L_SHRTI_WIDTH 1 #define MADERA_HP1L_SHRTO 0x0001 #define MADERA_HP1L_SHRTO_MASK 0x0001 #define MADERA_HP1L_SHRTO_SHIFT 0 -#define MADERA_HP1L_SHRTO_WIDTH 1 /* (0x0226) - HP Ctrl 1R */ #define MADERA_RMV_SHRT_HP1R 0x4000 #define MADERA_RMV_SHRT_HP1R_MASK 0x4000 #define MADERA_RMV_SHRT_HP1R_SHIFT 14 -#define MADERA_RMV_SHRT_HP1R_WIDTH 1 #define MADERA_HP1R_FLWR 0x0004 #define MADERA_HP1R_FLWR_MASK 0x0004 #define MADERA_HP1R_FLWR_SHIFT 2 -#define MADERA_HP1R_FLWR_WIDTH 1 #define MADERA_HP1R_SHRTI 0x0002 #define MADERA_HP1R_SHRTI_MASK 0x0002 #define MADERA_HP1R_SHRTI_SHIFT 1 -#define MADERA_HP1R_SHRTI_WIDTH 1 #define MADERA_HP1R_SHRTO 0x0001 #define MADERA_HP1R_SHRTO_MASK 0x0001 #define MADERA_HP1R_SHRTO_SHIFT 0 -#define MADERA_HP1R_SHRTO_WIDTH 1 /* (0x0293) Accessory_Detect_Mode_1 */ #define MADERA_ACCDET_SRC 0x2000 #define MADERA_ACCDET_SRC_MASK 0x2000 #define MADERA_ACCDET_SRC_SHIFT 13 -#define MADERA_ACCDET_SRC_WIDTH 1 #define MADERA_ACCDET_POLARITY_INV_ENA 0x0080 #define MADERA_ACCDET_POLARITY_INV_ENA_MASK 0x0080 #define MADERA_ACCDET_POLARITY_INV_ENA_SHIFT 7 -#define MADERA_ACCDET_POLARITY_INV_ENA_WIDTH 1 #define MADERA_ACCDET_MODE_MASK 0x0007 #define MADERA_ACCDET_MODE_SHIFT 0 -#define MADERA_ACCDET_MODE_WIDTH 3 /* (0x0299) Headphone_Detect_0 */ #define MADERA_HPD_GND_SEL 0x0007 #define MADERA_HPD_GND_SEL_MASK 0x0007 #define MADERA_HPD_GND_SEL_SHIFT 0 -#define MADERA_HPD_GND_SEL_WIDTH 3 #define MADERA_HPD_SENSE_SEL 0x00F0 #define MADERA_HPD_SENSE_SEL_MASK 0x00F0 #define MADERA_HPD_SENSE_SEL_SHIFT 4 -#define MADERA_HPD_SENSE_SEL_WIDTH 4 #define MADERA_HPD_FRC_SEL 0x0F00 #define MADERA_HPD_FRC_SEL_MASK 0x0F00 #define MADERA_HPD_FRC_SEL_SHIFT 8 -#define MADERA_HPD_FRC_SEL_WIDTH 4 #define MADERA_HPD_OUT_SEL 0x7000 #define MADERA_HPD_OUT_SEL_MASK 0x7000 #define MADERA_HPD_OUT_SEL_SHIFT 12 -#define MADERA_HPD_OUT_SEL_WIDTH 3 #define MADERA_HPD_OVD_ENA_SEL 0x8000 #define MADERA_HPD_OVD_ENA_SEL_MASK 0x8000 #define MADERA_HPD_OVD_ENA_SEL_SHIFT 15 -#define MADERA_HPD_OVD_ENA_SEL_WIDTH 1 /* (0x029B) Headphone_Detect_1 */ #define MADERA_HP_IMPEDANCE_RANGE_MASK 0x0600 #define MADERA_HP_IMPEDANCE_RANGE_SHIFT 9 -#define MADERA_HP_IMPEDANCE_RANGE_WIDTH 2 #define MADERA_HP_STEP_SIZE 0x0100 #define MADERA_HP_STEP_SIZE_MASK 0x0100 #define MADERA_HP_STEP_SIZE_SHIFT 8 -#define MADERA_HP_STEP_SIZE_WIDTH 1 #define MADERA_HP_CLK_DIV_MASK 0x0018 #define MADERA_HP_CLK_DIV_SHIFT 3 -#define MADERA_HP_CLK_DIV_WIDTH 2 #define MADERA_HP_RATE_MASK 0x0006 #define MADERA_HP_RATE_SHIFT 1 -#define MADERA_HP_RATE_WIDTH 2 #define MADERA_HP_POLL 0x0001 #define MADERA_HP_POLL_MASK 0x0001 #define MADERA_HP_POLL_SHIFT 0 -#define MADERA_HP_POLL_WIDTH 1 /* (0x029C) Headphone_Detect_2 */ #define MADERA_HP_DONE_MASK 0x8000 #define MADERA_HP_DONE_SHIFT 15 -#define MADERA_HP_DONE_WIDTH 1 #define MADERA_HP_LVL_MASK 0x7FFF #define MADERA_HP_LVL_SHIFT 0 -#define MADERA_HP_LVL_WIDTH 15 /* (0x029D) Headphone_Detect_3 */ #define MADERA_HP_DACVAL_MASK 0x03FF #define MADERA_HP_DACVAL_SHIFT 0 -#define MADERA_HP_DACVAL_WIDTH 10 /* (0x029F) - Headphone Detect 5 */ #define MADERA_HP_DACVAL_DOWN_MASK 0x03FF #define MADERA_HP_DACVAL_DOWN_SHIFT 0 -#define MADERA_HP_DACVAL_DOWN_WIDTH 10 /* (0x02A2) Mic_Detect_1_Control_0 */ #define MADERA_MICD1_GND_MASK 0x0007 #define MADERA_MICD1_GND_SHIFT 0 -#define MADERA_MICD1_GND_WIDTH 3 #define MADERA_MICD1_SENSE_MASK 0x00F0 #define MADERA_MICD1_SENSE_SHIFT 4 -#define MADERA_MICD1_SENSE_WIDTH 4 #define MADERA_MICD1_ADC_MODE_MASK 0x8000 #define MADERA_MICD1_ADC_MODE_SHIFT 15 -#define MADERA_MICD1_ADC_MODE_WIDTH 1 /* (0x02A3) Mic_Detect_1_Control_1 */ #define MADERA_MICD_BIAS_STARTTIME_MASK 0xF000 #define MADERA_MICD_BIAS_STARTTIME_SHIFT 12 -#define MADERA_MICD_BIAS_STARTTIME_WIDTH 4 #define MADERA_MICD_RATE_MASK 0x0F00 #define MADERA_MICD_RATE_SHIFT 8 -#define MADERA_MICD_RATE_WIDTH 4 #define MADERA_MICD_BIAS_SRC_MASK 0x00F0 #define MADERA_MICD_BIAS_SRC_SHIFT 4 -#define MADERA_MICD_BIAS_SRC_WIDTH 4 #define MADERA_MICD_DBTIME 0x0002 #define MADERA_MICD_DBTIME_MASK 0x0002 #define MADERA_MICD_DBTIME_SHIFT 1 -#define MADERA_MICD_DBTIME_WIDTH 1 #define MADERA_MICD_ENA 0x0001 #define MADERA_MICD_ENA_MASK 0x0001 #define MADERA_MICD_ENA_SHIFT 0 -#define MADERA_MICD_ENA_WIDTH 1 /* (0x02A4) Mic_Detect_1_Control_2 */ #define MADERA_MICD_LVL_SEL_MASK 0x00FF #define MADERA_MICD_LVL_SEL_SHIFT 0 -#define MADERA_MICD_LVL_SEL_WIDTH 8 /* (0x02A5) Mic_Detect_1_Control_3 */ #define MADERA_MICD_LVL_0 0x0004 @@ -1859,1746 +1731,1341 @@ #define MADERA_MICD_LVL_8 0x0400 #define MADERA_MICD_LVL_MASK 0x07FC #define MADERA_MICD_LVL_SHIFT 2 -#define MADERA_MICD_LVL_WIDTH 9 #define MADERA_MICD_VALID 0x0002 #define MADERA_MICD_VALID_MASK 0x0002 #define MADERA_MICD_VALID_SHIFT 1 -#define MADERA_MICD_VALID_WIDTH 1 #define MADERA_MICD_STS 0x0001 #define MADERA_MICD_STS_MASK 0x0001 #define MADERA_MICD_STS_SHIFT 0 -#define MADERA_MICD_STS_WIDTH 1 /* (0x02AB) Mic_Detect_1_Control_4 */ #define MADERA_MICDET_ADCVAL_DIFF_MASK 0xFF00 #define MADERA_MICDET_ADCVAL_DIFF_SHIFT 8 -#define MADERA_MICDET_ADCVAL_DIFF_WIDTH 8 #define MADERA_MICDET_ADCVAL_MASK 0x007F #define MADERA_MICDET_ADCVAL_SHIFT 0 -#define MADERA_MICDET_ADCVAL_WIDTH 7 /* (0x02C6) Micd_Clamp_control */ #define MADERA_MICD_CLAMP_OVD 0x0010 #define MADERA_MICD_CLAMP_OVD_MASK 0x0010 #define MADERA_MICD_CLAMP_OVD_SHIFT 4 -#define MADERA_MICD_CLAMP_OVD_WIDTH 1 #define MADERA_MICD_CLAMP_MODE_MASK 0x000F #define MADERA_MICD_CLAMP_MODE_SHIFT 0 -#define MADERA_MICD_CLAMP_MODE_WIDTH 4 /* (0x02C8) GP_Switch_1 */ #define MADERA_SW2_MODE_MASK 0x000C #define MADERA_SW2_MODE_SHIFT 2 -#define MADERA_SW2_MODE_WIDTH 2 #define MADERA_SW1_MODE_MASK 0x0003 #define MADERA_SW1_MODE_SHIFT 0 -#define MADERA_SW1_MODE_WIDTH 2 /* (0x02D3) Jack_detect_analogue */ #define MADERA_JD2_ENA 0x0002 #define MADERA_JD2_ENA_MASK 0x0002 #define MADERA_JD2_ENA_SHIFT 1 -#define MADERA_JD2_ENA_WIDTH 1 #define MADERA_JD1_ENA 0x0001 #define MADERA_JD1_ENA_MASK 0x0001 #define MADERA_JD1_ENA_SHIFT 0 -#define MADERA_JD1_ENA_WIDTH 1 /* (0x0300) Input_Enables */ #define MADERA_IN6L_ENA 0x0800 #define MADERA_IN6L_ENA_MASK 0x0800 #define MADERA_IN6L_ENA_SHIFT 11 -#define MADERA_IN6L_ENA_WIDTH 1 #define MADERA_IN6R_ENA 0x0400 #define MADERA_IN6R_ENA_MASK 0x0400 #define MADERA_IN6R_ENA_SHIFT 10 -#define MADERA_IN6R_ENA_WIDTH 1 #define MADERA_IN5L_ENA 0x0200 #define MADERA_IN5L_ENA_MASK 0x0200 #define MADERA_IN5L_ENA_SHIFT 9 -#define MADERA_IN5L_ENA_WIDTH 1 #define MADERA_IN5R_ENA 0x0100 #define MADERA_IN5R_ENA_MASK 0x0100 #define MADERA_IN5R_ENA_SHIFT 8 -#define MADERA_IN5R_ENA_WIDTH 1 #define MADERA_IN4L_ENA 0x0080 #define MADERA_IN4L_ENA_MASK 0x0080 #define MADERA_IN4L_ENA_SHIFT 7 -#define MADERA_IN4L_ENA_WIDTH 1 #define MADERA_IN4R_ENA 0x0040 #define MADERA_IN4R_ENA_MASK 0x0040 #define MADERA_IN4R_ENA_SHIFT 6 -#define MADERA_IN4R_ENA_WIDTH 1 #define MADERA_IN3L_ENA 0x0020 #define MADERA_IN3L_ENA_MASK 0x0020 #define MADERA_IN3L_ENA_SHIFT 5 -#define MADERA_IN3L_ENA_WIDTH 1 #define MADERA_IN3R_ENA 0x0010 #define MADERA_IN3R_ENA_MASK 0x0010 #define MADERA_IN3R_ENA_SHIFT 4 -#define MADERA_IN3R_ENA_WIDTH 1 #define MADERA_IN2L_ENA 0x0008 #define MADERA_IN2L_ENA_MASK 0x0008 #define MADERA_IN2L_ENA_SHIFT 3 -#define MADERA_IN2L_ENA_WIDTH 1 #define MADERA_IN2R_ENA 0x0004 #define MADERA_IN2R_ENA_MASK 0x0004 #define MADERA_IN2R_ENA_SHIFT 2 -#define MADERA_IN2R_ENA_WIDTH 1 #define MADERA_IN1L_ENA 0x0002 #define MADERA_IN1L_ENA_MASK 0x0002 #define MADERA_IN1L_ENA_SHIFT 1 -#define MADERA_IN1L_ENA_WIDTH 1 #define MADERA_IN1R_ENA 0x0001 #define MADERA_IN1R_ENA_MASK 0x0001 #define MADERA_IN1R_ENA_SHIFT 0 -#define MADERA_IN1R_ENA_WIDTH 1 /* (0x0308) Input_Rate */ #define MADERA_IN_RATE_MASK 0xF800 #define MADERA_IN_RATE_SHIFT 11 -#define MADERA_IN_RATE_WIDTH 5 #define MADERA_IN_MODE_MASK 0x0400 #define MADERA_IN_MODE_SHIFT 10 -#define MADERA_IN_MODE_WIDTH 1 /* (0x0309) Input_Volume_Ramp */ #define MADERA_IN_VD_RAMP_MASK 0x0070 #define MADERA_IN_VD_RAMP_SHIFT 4 -#define MADERA_IN_VD_RAMP_WIDTH 3 #define MADERA_IN_VI_RAMP_MASK 0x0007 #define MADERA_IN_VI_RAMP_SHIFT 0 -#define MADERA_IN_VI_RAMP_WIDTH 3 /* (0x030C) HPF_Control */ #define MADERA_IN_HPF_CUT_MASK 0x0007 #define MADERA_IN_HPF_CUT_SHIFT 0 -#define MADERA_IN_HPF_CUT_WIDTH 3 /* (0x0310) IN1L_Control */ #define MADERA_IN1L_HPF_MASK 0x8000 #define MADERA_IN1L_HPF_SHIFT 15 -#define MADERA_IN1L_HPF_WIDTH 1 #define MADERA_IN1_DMIC_SUP_MASK 0x1800 #define MADERA_IN1_DMIC_SUP_SHIFT 11 -#define MADERA_IN1_DMIC_SUP_WIDTH 2 #define MADERA_IN1_MODE_MASK 0x0400 #define MADERA_IN1_MODE_SHIFT 10 -#define MADERA_IN1_MODE_WIDTH 1 #define MADERA_IN1L_PGA_VOL_MASK 0x00FE #define MADERA_IN1L_PGA_VOL_SHIFT 1 -#define MADERA_IN1L_PGA_VOL_WIDTH 7 /* (0x0311) ADC_Digital_Volume_1L */ #define MADERA_IN1L_SRC_MASK 0x4000 #define MADERA_IN1L_SRC_SHIFT 14 -#define MADERA_IN1L_SRC_WIDTH 1 #define MADERA_IN1L_SRC_SE_MASK 0x2000 #define MADERA_IN1L_SRC_SE_SHIFT 13 -#define MADERA_IN1L_SRC_SE_WIDTH 1 #define MADERA_IN1L_LP_MODE 0x0800 #define MADERA_IN1L_LP_MODE_MASK 0x0800 #define MADERA_IN1L_LP_MODE_SHIFT 11 -#define MADERA_IN1L_LP_MODE_WIDTH 1 #define MADERA_IN_VU 0x0200 #define MADERA_IN_VU_MASK 0x0200 #define MADERA_IN_VU_SHIFT 9 -#define MADERA_IN_VU_WIDTH 1 #define MADERA_IN1L_MUTE 0x0100 #define MADERA_IN1L_MUTE_MASK 0x0100 #define MADERA_IN1L_MUTE_SHIFT 8 -#define MADERA_IN1L_MUTE_WIDTH 1 #define MADERA_IN1L_DIG_VOL_MASK 0x00FF #define MADERA_IN1L_DIG_VOL_SHIFT 0 -#define MADERA_IN1L_DIG_VOL_WIDTH 8 /* (0x0312) DMIC1L_Control */ #define MADERA_IN1_OSR_MASK 0x0700 #define MADERA_IN1_OSR_SHIFT 8 -#define MADERA_IN1_OSR_WIDTH 3 /* (0x0313) IN1L_Rate_Control */ #define MADERA_IN1L_RATE_MASK 0xF800 #define MADERA_IN1L_RATE_SHIFT 11 -#define MADERA_IN1L_RATE_WIDTH 5 /* (0x0314) IN1R_Control */ #define MADERA_IN1R_HPF_MASK 0x8000 #define MADERA_IN1R_HPF_SHIFT 15 -#define MADERA_IN1R_HPF_WIDTH 1 #define MADERA_IN1R_PGA_VOL_MASK 0x00FE #define MADERA_IN1R_PGA_VOL_SHIFT 1 -#define MADERA_IN1R_PGA_VOL_WIDTH 7 #define MADERA_IN1_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN1_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN1_DMICCLK_SRC_WIDTH 2 /* (0x0315) ADC_Digital_Volume_1R */ #define MADERA_IN1R_SRC_MASK 0x4000 #define MADERA_IN1R_SRC_SHIFT 14 -#define MADERA_IN1R_SRC_WIDTH 1 #define MADERA_IN1R_SRC_SE_MASK 0x2000 #define MADERA_IN1R_SRC_SE_SHIFT 13 -#define MADERA_IN1R_SRC_SE_WIDTH 1 #define MADERA_IN1R_LP_MODE 0x0800 #define MADERA_IN1R_LP_MODE_MASK 0x0800 #define MADERA_IN1R_LP_MODE_SHIFT 11 -#define MADERA_IN1R_LP_MODE_WIDTH 1 #define MADERA_IN1R_MUTE 0x0100 #define MADERA_IN1R_MUTE_MASK 0x0100 #define MADERA_IN1R_MUTE_SHIFT 8 -#define MADERA_IN1R_MUTE_WIDTH 1 #define MADERA_IN1R_DIG_VOL_MASK 0x00FF #define MADERA_IN1R_DIG_VOL_SHIFT 0 -#define MADERA_IN1R_DIG_VOL_WIDTH 8 /* (0x0317) IN1R_Rate_Control */ #define MADERA_IN1R_RATE_MASK 0xF800 #define MADERA_IN1R_RATE_SHIFT 11 -#define MADERA_IN1R_RATE_WIDTH 5 /* (0x0318) IN2L_Control */ #define MADERA_IN2L_HPF_MASK 0x8000 #define MADERA_IN2L_HPF_SHIFT 15 -#define MADERA_IN2L_HPF_WIDTH 1 #define MADERA_IN2_DMIC_SUP_MASK 0x1800 #define MADERA_IN2_DMIC_SUP_SHIFT 11 -#define MADERA_IN2_DMIC_SUP_WIDTH 2 #define MADERA_IN2_MODE_MASK 0x0400 #define MADERA_IN2_MODE_SHIFT 10 -#define MADERA_IN2_MODE_WIDTH 1 #define MADERA_IN2L_PGA_VOL_MASK 0x00FE #define MADERA_IN2L_PGA_VOL_SHIFT 1 -#define MADERA_IN2L_PGA_VOL_WIDTH 7 /* (0x0319) ADC_Digital_Volume_2L */ #define MADERA_IN2L_SRC_MASK 0x4000 #define MADERA_IN2L_SRC_SHIFT 14 -#define MADERA_IN2L_SRC_WIDTH 1 #define MADERA_IN2L_SRC_SE_MASK 0x2000 #define MADERA_IN2L_SRC_SE_SHIFT 13 -#define MADERA_IN2L_SRC_SE_WIDTH 1 #define MADERA_IN2L_LP_MODE 0x0800 #define MADERA_IN2L_LP_MODE_MASK 0x0800 #define MADERA_IN2L_LP_MODE_SHIFT 11 -#define MADERA_IN2L_LP_MODE_WIDTH 1 #define MADERA_IN2L_MUTE 0x0100 #define MADERA_IN2L_MUTE_MASK 0x0100 #define MADERA_IN2L_MUTE_SHIFT 8 -#define MADERA_IN2L_MUTE_WIDTH 1 #define MADERA_IN2L_DIG_VOL_MASK 0x00FF #define MADERA_IN2L_DIG_VOL_SHIFT 0 -#define MADERA_IN2L_DIG_VOL_WIDTH 8 /* (0x031A) DMIC2L_Control */ #define MADERA_IN2_OSR_MASK 0x0700 #define MADERA_IN2_OSR_SHIFT 8 -#define MADERA_IN2_OSR_WIDTH 3 /* (0x031C) IN2R_Control */ #define MADERA_IN2R_HPF_MASK 0x8000 #define MADERA_IN2R_HPF_SHIFT 15 -#define MADERA_IN2R_HPF_WIDTH 1 #define MADERA_IN2R_PGA_VOL_MASK 0x00FE #define MADERA_IN2R_PGA_VOL_SHIFT 1 -#define MADERA_IN2R_PGA_VOL_WIDTH 7 #define MADERA_IN2_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN2_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN2_DMICCLK_SRC_WIDTH 2 /* (0x031D) ADC_Digital_Volume_2R */ #define MADERA_IN2R_SRC_MASK 0x4000 #define MADERA_IN2R_SRC_SHIFT 14 -#define MADERA_IN2R_SRC_WIDTH 1 #define MADERA_IN2R_SRC_SE_MASK 0x2000 #define MADERA_IN2R_SRC_SE_SHIFT 13 -#define MADERA_IN2R_SRC_SE_WIDTH 1 #define MADERA_IN2R_LP_MODE 0x0800 #define MADERA_IN2R_LP_MODE_MASK 0x0800 #define MADERA_IN2R_LP_MODE_SHIFT 11 -#define MADERA_IN2R_LP_MODE_WIDTH 1 #define MADERA_IN2R_MUTE 0x0100 #define MADERA_IN2R_MUTE_MASK 0x0100 #define MADERA_IN2R_MUTE_SHIFT 8 -#define MADERA_IN2R_MUTE_WIDTH 1 #define MADERA_IN2R_DIG_VOL_MASK 0x00FF #define MADERA_IN2R_DIG_VOL_SHIFT 0 -#define MADERA_IN2R_DIG_VOL_WIDTH 8 /* (0x0320) IN3L_Control */ #define MADERA_IN3L_HPF_MASK 0x8000 #define MADERA_IN3L_HPF_SHIFT 15 -#define MADERA_IN3L_HPF_WIDTH 1 #define MADERA_IN3_DMIC_SUP_MASK 0x1800 #define MADERA_IN3_DMIC_SUP_SHIFT 11 -#define MADERA_IN3_DMIC_SUP_WIDTH 2 #define MADERA_IN3_MODE_MASK 0x0400 #define MADERA_IN3_MODE_SHIFT 10 -#define MADERA_IN3_MODE_WIDTH 1 #define MADERA_IN3L_PGA_VOL_MASK 0x00FE #define MADERA_IN3L_PGA_VOL_SHIFT 1 -#define MADERA_IN3L_PGA_VOL_WIDTH 7 /* (0x0321) ADC_Digital_Volume_3L */ #define MADERA_IN3L_MUTE 0x0100 #define MADERA_IN3L_MUTE_MASK 0x0100 #define MADERA_IN3L_MUTE_SHIFT 8 -#define MADERA_IN3L_MUTE_WIDTH 1 #define MADERA_IN3L_DIG_VOL_MASK 0x00FF #define MADERA_IN3L_DIG_VOL_SHIFT 0 -#define MADERA_IN3L_DIG_VOL_WIDTH 8 /* (0x0322) DMIC3L_Control */ #define MADERA_IN3_OSR_MASK 0x0700 #define MADERA_IN3_OSR_SHIFT 8 -#define MADERA_IN3_OSR_WIDTH 3 /* (0x0324) IN3R_Control */ #define MADERA_IN3R_HPF_MASK 0x8000 #define MADERA_IN3R_HPF_SHIFT 15 -#define MADERA_IN3R_HPF_WIDTH 1 #define MADERA_IN3R_PGA_VOL_MASK 0x00FE #define MADERA_IN3R_PGA_VOL_SHIFT 1 -#define MADERA_IN3R_PGA_VOL_WIDTH 7 #define MADERA_IN3_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN3_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN3_DMICCLK_SRC_WIDTH 2 /* (0x0325) ADC_Digital_Volume_3R */ #define MADERA_IN3R_MUTE 0x0100 #define MADERA_IN3R_MUTE_MASK 0x0100 #define MADERA_IN3R_MUTE_SHIFT 8 -#define MADERA_IN3R_MUTE_WIDTH 1 #define MADERA_IN3R_DIG_VOL_MASK 0x00FF #define MADERA_IN3R_DIG_VOL_SHIFT 0 -#define MADERA_IN3R_DIG_VOL_WIDTH 8 /* (0x0328) IN4L_Control */ #define MADERA_IN4L_HPF_MASK 0x8000 #define MADERA_IN4L_HPF_SHIFT 15 -#define MADERA_IN4L_HPF_WIDTH 1 #define MADERA_IN4_DMIC_SUP_MASK 0x1800 #define MADERA_IN4_DMIC_SUP_SHIFT 11 -#define MADERA_IN4_DMIC_SUP_WIDTH 2 /* (0x0329) ADC_Digital_Volume_4L */ #define MADERA_IN4L_MUTE 0x0100 #define MADERA_IN4L_MUTE_MASK 0x0100 #define MADERA_IN4L_MUTE_SHIFT 8 -#define MADERA_IN4L_MUTE_WIDTH 1 #define MADERA_IN4L_DIG_VOL_MASK 0x00FF #define MADERA_IN4L_DIG_VOL_SHIFT 0 -#define MADERA_IN4L_DIG_VOL_WIDTH 8 /* (0x032A) DMIC4L_Control */ #define MADERA_IN4_OSR_MASK 0x0700 #define MADERA_IN4_OSR_SHIFT 8 -#define MADERA_IN4_OSR_WIDTH 3 /* (0x032C) IN4R_Control */ #define MADERA_IN4R_HPF_MASK 0x8000 #define MADERA_IN4R_HPF_SHIFT 15 -#define MADERA_IN4R_HPF_WIDTH 1 #define MADERA_IN4_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN4_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN4_DMICCLK_SRC_WIDTH 2 /* (0x032D) ADC_Digital_Volume_4R */ #define MADERA_IN4R_MUTE 0x0100 #define MADERA_IN4R_MUTE_MASK 0x0100 #define MADERA_IN4R_MUTE_SHIFT 8 -#define MADERA_IN4R_MUTE_WIDTH 1 #define MADERA_IN4R_DIG_VOL_MASK 0x00FF #define MADERA_IN4R_DIG_VOL_SHIFT 0 -#define MADERA_IN4R_DIG_VOL_WIDTH 8 /* (0x0330) IN5L_Control */ #define MADERA_IN5L_HPF_MASK 0x8000 #define MADERA_IN5L_HPF_SHIFT 15 -#define MADERA_IN5L_HPF_WIDTH 1 #define MADERA_IN5_DMIC_SUP_MASK 0x1800 #define MADERA_IN5_DMIC_SUP_SHIFT 11 -#define MADERA_IN5_DMIC_SUP_WIDTH 2 /* (0x0331) ADC_Digital_Volume_5L */ #define MADERA_IN5L_MUTE 0x0100 #define MADERA_IN5L_MUTE_MASK 0x0100 #define MADERA_IN5L_MUTE_SHIFT 8 -#define MADERA_IN5L_MUTE_WIDTH 1 #define MADERA_IN5L_DIG_VOL_MASK 0x00FF #define MADERA_IN5L_DIG_VOL_SHIFT 0 -#define MADERA_IN5L_DIG_VOL_WIDTH 8 /* (0x0332) DMIC5L_Control */ #define MADERA_IN5_OSR_MASK 0x0700 #define MADERA_IN5_OSR_SHIFT 8 -#define MADERA_IN5_OSR_WIDTH 3 /* (0x0334) IN5R_Control */ #define MADERA_IN5R_HPF_MASK 0x8000 #define MADERA_IN5R_HPF_SHIFT 15 -#define MADERA_IN5R_HPF_WIDTH 1 #define MADERA_IN5_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN5_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN5_DMICCLK_SRC_WIDTH 2 /* (0x0335) ADC_Digital_Volume_5R */ #define MADERA_IN5R_MUTE 0x0100 #define MADERA_IN5R_MUTE_MASK 0x0100 #define MADERA_IN5R_MUTE_SHIFT 8 -#define MADERA_IN5R_MUTE_WIDTH 1 #define MADERA_IN5R_DIG_VOL_MASK 0x00FF #define MADERA_IN5R_DIG_VOL_SHIFT 0 -#define MADERA_IN5R_DIG_VOL_WIDTH 8 /* (0x0338) IN6L_Control */ #define MADERA_IN6L_HPF_MASK 0x8000 #define MADERA_IN6L_HPF_SHIFT 15 -#define MADERA_IN6L_HPF_WIDTH 1 #define MADERA_IN6_DMIC_SUP_MASK 0x1800 #define MADERA_IN6_DMIC_SUP_SHIFT 11 -#define MADERA_IN6_DMIC_SUP_WIDTH 2 /* (0x0339) ADC_Digital_Volume_6L */ #define MADERA_IN6L_MUTE 0x0100 #define MADERA_IN6L_MUTE_MASK 0x0100 #define MADERA_IN6L_MUTE_SHIFT 8 -#define MADERA_IN6L_MUTE_WIDTH 1 #define MADERA_IN6L_DIG_VOL_MASK 0x00FF #define MADERA_IN6L_DIG_VOL_SHIFT 0 -#define MADERA_IN6L_DIG_VOL_WIDTH 8 /* (0x033A) DMIC6L_Control */ #define MADERA_IN6_OSR_MASK 0x0700 #define MADERA_IN6_OSR_SHIFT 8 -#define MADERA_IN6_OSR_WIDTH 3 /* (0x033C) IN6R_Control */ #define MADERA_IN6R_HPF_MASK 0x8000 #define MADERA_IN6R_HPF_SHIFT 15 -#define MADERA_IN6R_HPF_WIDTH 1 /* (0x033D) ADC_Digital_Volume_6R */ #define MADERA_IN6R_MUTE 0x0100 #define MADERA_IN6R_MUTE_MASK 0x0100 #define MADERA_IN6R_MUTE_SHIFT 8 -#define MADERA_IN6R_MUTE_WIDTH 1 #define MADERA_IN6R_DIG_VOL_MASK 0x00FF #define MADERA_IN6R_DIG_VOL_SHIFT 0 -#define MADERA_IN6R_DIG_VOL_WIDTH 8 /* (0x033E) DMIC6R_Control */ #define MADERA_IN6_DMICCLK_SRC_MASK 0x1800 #define MADERA_IN6_DMICCLK_SRC_SHIFT 11 -#define MADERA_IN6_DMICCLK_SRC_WIDTH 2 /* (0x0400) Output_Enables_1 */ #define MADERA_EP_SEL 0x8000 #define MADERA_EP_SEL_MASK 0x8000 #define MADERA_EP_SEL_SHIFT 15 -#define MADERA_EP_SEL_WIDTH 1 #define MADERA_OUT6L_ENA 0x0800 #define MADERA_OUT6L_ENA_MASK 0x0800 #define MADERA_OUT6L_ENA_SHIFT 11 -#define MADERA_OUT6L_ENA_WIDTH 1 #define MADERA_OUT6R_ENA 0x0400 #define MADERA_OUT6R_ENA_MASK 0x0400 #define MADERA_OUT6R_ENA_SHIFT 10 -#define MADERA_OUT6R_ENA_WIDTH 1 #define MADERA_OUT5L_ENA 0x0200 #define MADERA_OUT5L_ENA_MASK 0x0200 #define MADERA_OUT5L_ENA_SHIFT 9 -#define MADERA_OUT5L_ENA_WIDTH 1 #define MADERA_OUT5R_ENA 0x0100 #define MADERA_OUT5R_ENA_MASK 0x0100 #define MADERA_OUT5R_ENA_SHIFT 8 -#define MADERA_OUT5R_ENA_WIDTH 1 #define MADERA_OUT4L_ENA 0x0080 #define MADERA_OUT4L_ENA_MASK 0x0080 #define MADERA_OUT4L_ENA_SHIFT 7 -#define MADERA_OUT4L_ENA_WIDTH 1 #define MADERA_OUT4R_ENA 0x0040 #define MADERA_OUT4R_ENA_MASK 0x0040 #define MADERA_OUT4R_ENA_SHIFT 6 -#define MADERA_OUT4R_ENA_WIDTH 1 #define MADERA_OUT3L_ENA 0x0020 #define MADERA_OUT3L_ENA_MASK 0x0020 #define MADERA_OUT3L_ENA_SHIFT 5 -#define MADERA_OUT3L_ENA_WIDTH 1 #define MADERA_OUT3R_ENA 0x0010 #define MADERA_OUT3R_ENA_MASK 0x0010 #define MADERA_OUT3R_ENA_SHIFT 4 -#define MADERA_OUT3R_ENA_WIDTH 1 #define MADERA_OUT2L_ENA 0x0008 #define MADERA_OUT2L_ENA_MASK 0x0008 #define MADERA_OUT2L_ENA_SHIFT 3 -#define MADERA_OUT2L_ENA_WIDTH 1 #define MADERA_OUT2R_ENA 0x0004 #define MADERA_OUT2R_ENA_MASK 0x0004 #define MADERA_OUT2R_ENA_SHIFT 2 -#define MADERA_OUT2R_ENA_WIDTH 1 #define MADERA_OUT1L_ENA 0x0002 #define MADERA_OUT1L_ENA_MASK 0x0002 #define MADERA_OUT1L_ENA_SHIFT 1 -#define MADERA_OUT1L_ENA_WIDTH 1 #define MADERA_OUT1R_ENA 0x0001 #define MADERA_OUT1R_ENA_MASK 0x0001 #define MADERA_OUT1R_ENA_SHIFT 0 -#define MADERA_OUT1R_ENA_WIDTH 1 /* (0x0408) Output_Rate_1 */ #define MADERA_CP_DAC_MODE_MASK 0x0040 #define MADERA_CP_DAC_MODE_SHIFT 6 -#define MADERA_CP_DAC_MODE_WIDTH 1 #define MADERA_OUT_EXT_CLK_DIV_MASK 0x0030 #define MADERA_OUT_EXT_CLK_DIV_SHIFT 4 -#define MADERA_OUT_EXT_CLK_DIV_WIDTH 2 #define MADERA_OUT_CLK_SRC_MASK 0x0007 #define MADERA_OUT_CLK_SRC_SHIFT 0 -#define MADERA_OUT_CLK_SRC_WIDTH 3 /* (0x0409) Output_Volume_Ramp */ #define MADERA_OUT_VD_RAMP_MASK 0x0070 #define MADERA_OUT_VD_RAMP_SHIFT 4 -#define MADERA_OUT_VD_RAMP_WIDTH 3 #define MADERA_OUT_VI_RAMP_MASK 0x0007 #define MADERA_OUT_VI_RAMP_SHIFT 0 -#define MADERA_OUT_VI_RAMP_WIDTH 3 /* (0x0410) Output_Path_Config_1L */ #define MADERA_OUT1_MONO 0x1000 #define MADERA_OUT1_MONO_MASK 0x1000 #define MADERA_OUT1_MONO_SHIFT 12 -#define MADERA_OUT1_MONO_WIDTH 1 #define MADERA_OUT1L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT1L_ANC_SRC_SHIFT 10 -#define MADERA_OUT1L_ANC_SRC_WIDTH 2 /* (0x0411) DAC_Digital_Volume_1L */ #define MADERA_OUT1L_VU 0x0200 #define MADERA_OUT1L_VU_MASK 0x0200 #define MADERA_OUT1L_VU_SHIFT 9 -#define MADERA_OUT1L_VU_WIDTH 1 #define MADERA_OUT1L_MUTE 0x0100 #define MADERA_OUT1L_MUTE_MASK 0x0100 #define MADERA_OUT1L_MUTE_SHIFT 8 -#define MADERA_OUT1L_MUTE_WIDTH 1 #define MADERA_OUT1L_VOL_MASK 0x00FF #define MADERA_OUT1L_VOL_SHIFT 0 -#define MADERA_OUT1L_VOL_WIDTH 8 /* (0x0412) Output_Path_Config_1 */ #define MADERA_HP1_GND_SEL_MASK 0x0007 #define MADERA_HP1_GND_SEL_SHIFT 0 -#define MADERA_HP1_GND_SEL_WIDTH 3 /* (0x0414) Output_Path_Config_1R */ #define MADERA_OUT1R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT1R_ANC_SRC_SHIFT 10 -#define MADERA_OUT1R_ANC_SRC_WIDTH 2 /* (0x0415) DAC_Digital_Volume_1R */ #define MADERA_OUT1R_MUTE 0x0100 #define MADERA_OUT1R_MUTE_MASK 0x0100 #define MADERA_OUT1R_MUTE_SHIFT 8 -#define MADERA_OUT1R_MUTE_WIDTH 1 #define MADERA_OUT1R_VOL_MASK 0x00FF #define MADERA_OUT1R_VOL_SHIFT 0 -#define MADERA_OUT1R_VOL_WIDTH 8 /* (0x0418) Output_Path_Config_2L */ #define MADERA_OUT2L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT2L_ANC_SRC_SHIFT 10 -#define MADERA_OUT2L_ANC_SRC_WIDTH 2 /* (0x0419) DAC_Digital_Volume_2L */ #define MADERA_OUT2L_MUTE 0x0100 #define MADERA_OUT2L_MUTE_MASK 0x0100 #define MADERA_OUT2L_MUTE_SHIFT 8 -#define MADERA_OUT2L_MUTE_WIDTH 1 #define MADERA_OUT2L_VOL_MASK 0x00FF #define MADERA_OUT2L_VOL_SHIFT 0 -#define MADERA_OUT2L_VOL_WIDTH 8 /* (0x041A) Output_Path_Config_2 */ #define MADERA_HP2_GND_SEL_MASK 0x0007 #define MADERA_HP2_GND_SEL_SHIFT 0 -#define MADERA_HP2_GND_SEL_WIDTH 3 /* (0x041C) Output_Path_Config_2R */ #define MADERA_OUT2R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT2R_ANC_SRC_SHIFT 10 -#define MADERA_OUT2R_ANC_SRC_WIDTH 2 /* (0x041D) DAC_Digital_Volume_2R */ #define MADERA_OUT2R_MUTE 0x0100 #define MADERA_OUT2R_MUTE_MASK 0x0100 #define MADERA_OUT2R_MUTE_SHIFT 8 -#define MADERA_OUT2R_MUTE_WIDTH 1 #define MADERA_OUT2R_VOL_MASK 0x00FF #define MADERA_OUT2R_VOL_SHIFT 0 -#define MADERA_OUT2R_VOL_WIDTH 8 /* (0x0420) Output_Path_Config_3L */ #define MADERA_OUT3L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT3L_ANC_SRC_SHIFT 10 -#define MADERA_OUT3L_ANC_SRC_WIDTH 2 /* (0x0421) DAC_Digital_Volume_3L */ #define MADERA_OUT3L_MUTE 0x0100 #define MADERA_OUT3L_MUTE_MASK 0x0100 #define MADERA_OUT3L_MUTE_SHIFT 8 -#define MADERA_OUT3L_MUTE_WIDTH 1 #define MADERA_OUT3L_VOL_MASK 0x00FF #define MADERA_OUT3L_VOL_SHIFT 0 -#define MADERA_OUT3L_VOL_WIDTH 8 /* (0x0424) Output_Path_Config_3R */ #define MADERA_OUT3R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT3R_ANC_SRC_SHIFT 10 -#define MADERA_OUT3R_ANC_SRC_WIDTH 2 /* (0x0425) DAC_Digital_Volume_3R */ #define MADERA_OUT3R_MUTE 0x0100 #define MADERA_OUT3R_MUTE_MASK 0x0100 #define MADERA_OUT3R_MUTE_SHIFT 8 -#define MADERA_OUT3R_MUTE_WIDTH 1 #define MADERA_OUT3R_VOL_MASK 0x00FF #define MADERA_OUT3R_VOL_SHIFT 0 -#define MADERA_OUT3R_VOL_WIDTH 8 /* (0x0428) Output_Path_Config_4L */ #define MADERA_OUT4L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT4L_ANC_SRC_SHIFT 10 -#define MADERA_OUT4L_ANC_SRC_WIDTH 2 /* (0x0429) DAC_Digital_Volume_4L */ #define MADERA_OUT4L_MUTE 0x0100 #define MADERA_OUT4L_MUTE_MASK 0x0100 #define MADERA_OUT4L_MUTE_SHIFT 8 -#define MADERA_OUT4L_MUTE_WIDTH 1 #define MADERA_OUT4L_VOL_MASK 0x00FF #define MADERA_OUT4L_VOL_SHIFT 0 -#define MADERA_OUT4L_VOL_WIDTH 8 /* (0x042C) Output_Path_Config_4R */ #define MADERA_OUT4R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT4R_ANC_SRC_SHIFT 10 -#define MADERA_OUT4R_ANC_SRC_WIDTH 2 /* (0x042D) DAC_Digital_Volume_4R */ #define MADERA_OUT4R_MUTE 0x0100 #define MADERA_OUT4R_MUTE_MASK 0x0100 #define MADERA_OUT4R_MUTE_SHIFT 8 -#define MADERA_OUT4R_MUTE_WIDTH 1 #define MADERA_OUT4R_VOL_MASK 0x00FF #define MADERA_OUT4R_VOL_SHIFT 0 -#define MADERA_OUT4R_VOL_WIDTH 8 /* (0x0430) Output_Path_Config_5L */ #define MADERA_OUT5_OSR 0x2000 #define MADERA_OUT5_OSR_MASK 0x2000 #define MADERA_OUT5_OSR_SHIFT 13 -#define MADERA_OUT5_OSR_WIDTH 1 #define MADERA_OUT5L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT5L_ANC_SRC_SHIFT 10 -#define MADERA_OUT5L_ANC_SRC_WIDTH 2 /* (0x0431) DAC_Digital_Volume_5L */ #define MADERA_OUT5L_MUTE 0x0100 #define MADERA_OUT5L_MUTE_MASK 0x0100 #define MADERA_OUT5L_MUTE_SHIFT 8 -#define MADERA_OUT5L_MUTE_WIDTH 1 #define MADERA_OUT5L_VOL_MASK 0x00FF #define MADERA_OUT5L_VOL_SHIFT 0 -#define MADERA_OUT5L_VOL_WIDTH 8 /* (0x0434) Output_Path_Config_5R */ #define MADERA_OUT5R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT5R_ANC_SRC_SHIFT 10 -#define MADERA_OUT5R_ANC_SRC_WIDTH 2 /* (0x0435) DAC_Digital_Volume_5R */ #define MADERA_OUT5R_MUTE 0x0100 #define MADERA_OUT5R_MUTE_MASK 0x0100 #define MADERA_OUT5R_MUTE_SHIFT 8 -#define MADERA_OUT5R_MUTE_WIDTH 1 #define MADERA_OUT5R_VOL_MASK 0x00FF #define MADERA_OUT5R_VOL_SHIFT 0 -#define MADERA_OUT5R_VOL_WIDTH 8 /* (0x0438) Output_Path_Config_6L */ #define MADERA_OUT6_OSR 0x2000 #define MADERA_OUT6_OSR_MASK 0x2000 #define MADERA_OUT6_OSR_SHIFT 13 -#define MADERA_OUT6_OSR_WIDTH 1 #define MADERA_OUT6L_ANC_SRC_MASK 0x0C00 #define MADERA_OUT6L_ANC_SRC_SHIFT 10 -#define MADERA_OUT6L_ANC_SRC_WIDTH 2 /* (0x0439) DAC_Digital_Volume_6L */ #define MADERA_OUT6L_MUTE 0x0100 #define MADERA_OUT6L_MUTE_MASK 0x0100 #define MADERA_OUT6L_MUTE_SHIFT 8 -#define MADERA_OUT6L_MUTE_WIDTH 1 #define MADERA_OUT6L_VOL_MASK 0x00FF #define MADERA_OUT6L_VOL_SHIFT 0 -#define MADERA_OUT6L_VOL_WIDTH 8 /* (0x043C) Output_Path_Config_6R */ #define MADERA_OUT6R_ANC_SRC_MASK 0x0C00 #define MADERA_OUT6R_ANC_SRC_SHIFT 10 -#define MADERA_OUT6R_ANC_SRC_WIDTH 2 /* (0x043D) DAC_Digital_Volume_6R */ #define MADERA_OUT6R_MUTE 0x0100 #define MADERA_OUT6R_MUTE_MASK 0x0100 #define MADERA_OUT6R_MUTE_SHIFT 8 -#define MADERA_OUT6R_MUTE_WIDTH 1 #define MADERA_OUT6R_VOL_MASK 0x00FF #define MADERA_OUT6R_VOL_SHIFT 0 -#define MADERA_OUT6R_VOL_WIDTH 8 /* (0x0450) - DAC AEC Control 1 */ #define MADERA_AEC1_LOOPBACK_SRC_MASK 0x003C #define MADERA_AEC1_LOOPBACK_SRC_SHIFT 2 -#define MADERA_AEC1_LOOPBACK_SRC_WIDTH 4 #define MADERA_AEC1_ENA_STS 0x0002 #define MADERA_AEC1_ENA_STS_MASK 0x0002 #define MADERA_AEC1_ENA_STS_SHIFT 1 -#define MADERA_AEC1_ENA_STS_WIDTH 1 #define MADERA_AEC1_LOOPBACK_ENA 0x0001 #define MADERA_AEC1_LOOPBACK_ENA_MASK 0x0001 #define MADERA_AEC1_LOOPBACK_ENA_SHIFT 0 -#define MADERA_AEC1_LOOPBACK_ENA_WIDTH 1 /* (0x0451) DAC_AEC_Control_2 */ #define MADERA_AEC2_LOOPBACK_SRC_MASK 0x003C #define MADERA_AEC2_LOOPBACK_SRC_SHIFT 2 -#define MADERA_AEC2_LOOPBACK_SRC_WIDTH 4 #define MADERA_AEC2_ENA_STS 0x0002 #define MADERA_AEC2_ENA_STS_MASK 0x0002 #define MADERA_AEC2_ENA_STS_SHIFT 1 -#define MADERA_AEC2_ENA_STS_WIDTH 1 #define MADERA_AEC2_LOOPBACK_ENA 0x0001 #define MADERA_AEC2_LOOPBACK_ENA_MASK 0x0001 #define MADERA_AEC2_LOOPBACK_ENA_SHIFT 0 -#define MADERA_AEC2_LOOPBACK_ENA_WIDTH 1 /* (0x0458) Noise_Gate_Control */ #define MADERA_NGATE_HOLD_MASK 0x0030 #define MADERA_NGATE_HOLD_SHIFT 4 -#define MADERA_NGATE_HOLD_WIDTH 2 #define MADERA_NGATE_THR_MASK 0x000E #define MADERA_NGATE_THR_SHIFT 1 -#define MADERA_NGATE_THR_WIDTH 3 #define MADERA_NGATE_ENA 0x0001 #define MADERA_NGATE_ENA_MASK 0x0001 #define MADERA_NGATE_ENA_SHIFT 0 -#define MADERA_NGATE_ENA_WIDTH 1 /* (0x0490) PDM_SPK1_CTRL_1 */ #define MADERA_SPK1R_MUTE 0x2000 #define MADERA_SPK1R_MUTE_MASK 0x2000 #define MADERA_SPK1R_MUTE_SHIFT 13 -#define MADERA_SPK1R_MUTE_WIDTH 1 #define MADERA_SPK1L_MUTE 0x1000 #define MADERA_SPK1L_MUTE_MASK 0x1000 #define MADERA_SPK1L_MUTE_SHIFT 12 -#define MADERA_SPK1L_MUTE_WIDTH 1 #define MADERA_SPK1_MUTE_ENDIAN 0x0100 #define MADERA_SPK1_MUTE_ENDIAN_MASK 0x0100 #define MADERA_SPK1_MUTE_ENDIAN_SHIFT 8 -#define MADERA_SPK1_MUTE_ENDIAN_WIDTH 1 #define MADERA_SPK1_MUTE_SEQ1_MASK 0x00FF #define MADERA_SPK1_MUTE_SEQ1_SHIFT 0 -#define MADERA_SPK1_MUTE_SEQ1_WIDTH 8 /* (0x0491) PDM_SPK1_CTRL_2 */ #define MADERA_SPK1_FMT 0x0001 #define MADERA_SPK1_FMT_MASK 0x0001 #define MADERA_SPK1_FMT_SHIFT 0 -#define MADERA_SPK1_FMT_WIDTH 1 /* (0x0492) PDM_SPK2_CTRL_1 */ #define MADERA_SPK2R_MUTE 0x2000 #define MADERA_SPK2R_MUTE_MASK 0x2000 #define MADERA_SPK2R_MUTE_SHIFT 13 -#define MADERA_SPK2R_MUTE_WIDTH 1 #define MADERA_SPK2L_MUTE 0x1000 #define MADERA_SPK2L_MUTE_MASK 0x1000 #define MADERA_SPK2L_MUTE_SHIFT 12 -#define MADERA_SPK2L_MUTE_WIDTH 1 /* (0x04A0) - HP1 Short Circuit Ctrl */ #define MADERA_HP1_SC_ENA 0x1000 #define MADERA_HP1_SC_ENA_MASK 0x1000 #define MADERA_HP1_SC_ENA_SHIFT 12 -#define MADERA_HP1_SC_ENA_WIDTH 1 /* (0x04A1) - HP2 Short Circuit Ctrl */ #define MADERA_HP2_SC_ENA 0x1000 #define MADERA_HP2_SC_ENA_MASK 0x1000 #define MADERA_HP2_SC_ENA_SHIFT 12 -#define MADERA_HP2_SC_ENA_WIDTH 1 /* (0x04A2) - HP3 Short Circuit Ctrl */ #define MADERA_HP3_SC_ENA 0x1000 #define MADERA_HP3_SC_ENA_MASK 0x1000 #define MADERA_HP3_SC_ENA_SHIFT 12 -#define MADERA_HP3_SC_ENA_WIDTH 1 /* (0x04A8) - HP_Test_Ctrl_5 */ #define MADERA_HP1L_ONEFLT 0x0100 #define MADERA_HP1L_ONEFLT_MASK 0x0100 #define MADERA_HP1L_ONEFLT_SHIFT 8 -#define MADERA_HP1L_ONEFLT_WIDTH 1 /* (0x04A9) - HP_Test_Ctrl_6 */ #define MADERA_HP1R_ONEFLT 0x0100 #define MADERA_HP1R_ONEFLT_MASK 0x0100 #define MADERA_HP1R_ONEFLT_SHIFT 8 -#define MADERA_HP1R_ONEFLT_WIDTH 1 /* (0x0500) AIF1_BCLK_Ctrl */ #define MADERA_AIF1_BCLK_INV 0x0080 #define MADERA_AIF1_BCLK_INV_MASK 0x0080 #define MADERA_AIF1_BCLK_INV_SHIFT 7 -#define MADERA_AIF1_BCLK_INV_WIDTH 1 #define MADERA_AIF1_BCLK_MSTR 0x0020 #define MADERA_AIF1_BCLK_MSTR_MASK 0x0020 #define MADERA_AIF1_BCLK_MSTR_SHIFT 5 -#define MADERA_AIF1_BCLK_MSTR_WIDTH 1 #define MADERA_AIF1_BCLK_FREQ_MASK 0x001F #define MADERA_AIF1_BCLK_FREQ_SHIFT 0 -#define MADERA_AIF1_BCLK_FREQ_WIDTH 5 /* (0x0501) AIF1_Tx_Pin_Ctrl */ #define MADERA_AIF1TX_LRCLK_SRC 0x0008 #define MADERA_AIF1TX_LRCLK_SRC_MASK 0x0008 #define MADERA_AIF1TX_LRCLK_SRC_SHIFT 3 -#define MADERA_AIF1TX_LRCLK_SRC_WIDTH 1 #define MADERA_AIF1TX_LRCLK_INV 0x0004 #define MADERA_AIF1TX_LRCLK_INV_MASK 0x0004 #define MADERA_AIF1TX_LRCLK_INV_SHIFT 2 -#define MADERA_AIF1TX_LRCLK_INV_WIDTH 1 #define MADERA_AIF1TX_LRCLK_MSTR 0x0001 #define MADERA_AIF1TX_LRCLK_MSTR_MASK 0x0001 #define MADERA_AIF1TX_LRCLK_MSTR_SHIFT 0 -#define MADERA_AIF1TX_LRCLK_MSTR_WIDTH 1 /* (0x0502) AIF1_Rx_Pin_Ctrl */ #define MADERA_AIF1RX_LRCLK_INV 0x0004 #define MADERA_AIF1RX_LRCLK_INV_MASK 0x0004 #define MADERA_AIF1RX_LRCLK_INV_SHIFT 2 -#define MADERA_AIF1RX_LRCLK_INV_WIDTH 1 #define MADERA_AIF1RX_LRCLK_FRC 0x0002 #define MADERA_AIF1RX_LRCLK_FRC_MASK 0x0002 #define MADERA_AIF1RX_LRCLK_FRC_SHIFT 1 -#define MADERA_AIF1RX_LRCLK_FRC_WIDTH 1 #define MADERA_AIF1RX_LRCLK_MSTR 0x0001 #define MADERA_AIF1RX_LRCLK_MSTR_MASK 0x0001 #define MADERA_AIF1RX_LRCLK_MSTR_SHIFT 0 -#define MADERA_AIF1RX_LRCLK_MSTR_WIDTH 1 /* (0x0503) AIF1_Rate_Ctrl */ #define MADERA_AIF1_RATE_MASK 0xF800 #define MADERA_AIF1_RATE_SHIFT 11 -#define MADERA_AIF1_RATE_WIDTH 5 #define MADERA_AIF1_TRI 0x0040 #define MADERA_AIF1_TRI_MASK 0x0040 #define MADERA_AIF1_TRI_SHIFT 6 -#define MADERA_AIF1_TRI_WIDTH 1 /* (0x0504) AIF1_Format */ #define MADERA_AIF1_FMT_MASK 0x0007 #define MADERA_AIF1_FMT_SHIFT 0 -#define MADERA_AIF1_FMT_WIDTH 3 /* (0x0506) AIF1_Rx_BCLK_Rate */ #define MADERA_AIF1RX_BCPF_MASK 0x1FFF #define MADERA_AIF1RX_BCPF_SHIFT 0 -#define MADERA_AIF1RX_BCPF_WIDTH 13 /* (0x0507) AIF1_Frame_Ctrl_1 */ #define MADERA_AIF1TX_WL_MASK 0x3F00 #define MADERA_AIF1TX_WL_SHIFT 8 -#define MADERA_AIF1TX_WL_WIDTH 6 #define MADERA_AIF1TX_SLOT_LEN_MASK 0x00FF #define MADERA_AIF1TX_SLOT_LEN_SHIFT 0 -#define MADERA_AIF1TX_SLOT_LEN_WIDTH 8 /* (0x0508) AIF1_Frame_Ctrl_2 */ #define MADERA_AIF1RX_WL_MASK 0x3F00 #define MADERA_AIF1RX_WL_SHIFT 8 -#define MADERA_AIF1RX_WL_WIDTH 6 #define MADERA_AIF1RX_SLOT_LEN_MASK 0x00FF #define MADERA_AIF1RX_SLOT_LEN_SHIFT 0 -#define MADERA_AIF1RX_SLOT_LEN_WIDTH 8 /* (0x0509) AIF1_Frame_Ctrl_3 */ #define MADERA_AIF1TX1_SLOT_MASK 0x003F #define MADERA_AIF1TX1_SLOT_SHIFT 0 -#define MADERA_AIF1TX1_SLOT_WIDTH 6 /* (0x0519) AIF1_Tx_Enables */ #define MADERA_AIF1TX8_ENA 0x0080 #define MADERA_AIF1TX8_ENA_MASK 0x0080 #define MADERA_AIF1TX8_ENA_SHIFT 7 -#define MADERA_AIF1TX8_ENA_WIDTH 1 #define MADERA_AIF1TX7_ENA 0x0040 #define MADERA_AIF1TX7_ENA_MASK 0x0040 #define MADERA_AIF1TX7_ENA_SHIFT 6 -#define MADERA_AIF1TX7_ENA_WIDTH 1 #define MADERA_AIF1TX6_ENA 0x0020 #define MADERA_AIF1TX6_ENA_MASK 0x0020 #define MADERA_AIF1TX6_ENA_SHIFT 5 -#define MADERA_AIF1TX6_ENA_WIDTH 1 #define MADERA_AIF1TX5_ENA 0x0010 #define MADERA_AIF1TX5_ENA_MASK 0x0010 #define MADERA_AIF1TX5_ENA_SHIFT 4 -#define MADERA_AIF1TX5_ENA_WIDTH 1 #define MADERA_AIF1TX4_ENA 0x0008 #define MADERA_AIF1TX4_ENA_MASK 0x0008 #define MADERA_AIF1TX4_ENA_SHIFT 3 -#define MADERA_AIF1TX4_ENA_WIDTH 1 #define MADERA_AIF1TX3_ENA 0x0004 #define MADERA_AIF1TX3_ENA_MASK 0x0004 #define MADERA_AIF1TX3_ENA_SHIFT 2 -#define MADERA_AIF1TX3_ENA_WIDTH 1 #define MADERA_AIF1TX2_ENA 0x0002 #define MADERA_AIF1TX2_ENA_MASK 0x0002 #define MADERA_AIF1TX2_ENA_SHIFT 1 -#define MADERA_AIF1TX2_ENA_WIDTH 1 #define MADERA_AIF1TX1_ENA 0x0001 #define MADERA_AIF1TX1_ENA_MASK 0x0001 #define MADERA_AIF1TX1_ENA_SHIFT 0 -#define MADERA_AIF1TX1_ENA_WIDTH 1 /* (0x051A) AIF1_Rx_Enables */ #define MADERA_AIF1RX8_ENA 0x0080 #define MADERA_AIF1RX8_ENA_MASK 0x0080 #define MADERA_AIF1RX8_ENA_SHIFT 7 -#define MADERA_AIF1RX8_ENA_WIDTH 1 #define MADERA_AIF1RX7_ENA 0x0040 #define MADERA_AIF1RX7_ENA_MASK 0x0040 #define MADERA_AIF1RX7_ENA_SHIFT 6 -#define MADERA_AIF1RX7_ENA_WIDTH 1 #define MADERA_AIF1RX6_ENA 0x0020 #define MADERA_AIF1RX6_ENA_MASK 0x0020 #define MADERA_AIF1RX6_ENA_SHIFT 5 -#define MADERA_AIF1RX6_ENA_WIDTH 1 #define MADERA_AIF1RX5_ENA 0x0010 #define MADERA_AIF1RX5_ENA_MASK 0x0010 #define MADERA_AIF1RX5_ENA_SHIFT 4 -#define MADERA_AIF1RX5_ENA_WIDTH 1 #define MADERA_AIF1RX4_ENA 0x0008 #define MADERA_AIF1RX4_ENA_MASK 0x0008 #define MADERA_AIF1RX4_ENA_SHIFT 3 -#define MADERA_AIF1RX4_ENA_WIDTH 1 #define MADERA_AIF1RX3_ENA 0x0004 #define MADERA_AIF1RX3_ENA_MASK 0x0004 #define MADERA_AIF1RX3_ENA_SHIFT 2 -#define MADERA_AIF1RX3_ENA_WIDTH 1 #define MADERA_AIF1RX2_ENA 0x0002 #define MADERA_AIF1RX2_ENA_MASK 0x0002 #define MADERA_AIF1RX2_ENA_SHIFT 1 -#define MADERA_AIF1RX2_ENA_WIDTH 1 #define MADERA_AIF1RX1_ENA 0x0001 #define MADERA_AIF1RX1_ENA_MASK 0x0001 #define MADERA_AIF1RX1_ENA_SHIFT 0 -#define MADERA_AIF1RX1_ENA_WIDTH 1 /* (0x0559) AIF2_Tx_Enables */ #define MADERA_AIF2TX8_ENA 0x0080 #define MADERA_AIF2TX8_ENA_MASK 0x0080 #define MADERA_AIF2TX8_ENA_SHIFT 7 -#define MADERA_AIF2TX8_ENA_WIDTH 1 #define MADERA_AIF2TX7_ENA 0x0040 #define MADERA_AIF2TX7_ENA_MASK 0x0040 #define MADERA_AIF2TX7_ENA_SHIFT 6 -#define MADERA_AIF2TX7_ENA_WIDTH 1 #define MADERA_AIF2TX6_ENA 0x0020 #define MADERA_AIF2TX6_ENA_MASK 0x0020 #define MADERA_AIF2TX6_ENA_SHIFT 5 -#define MADERA_AIF2TX6_ENA_WIDTH 1 #define MADERA_AIF2TX5_ENA 0x0010 #define MADERA_AIF2TX5_ENA_MASK 0x0010 #define MADERA_AIF2TX5_ENA_SHIFT 4 -#define MADERA_AIF2TX5_ENA_WIDTH 1 #define MADERA_AIF2TX4_ENA 0x0008 #define MADERA_AIF2TX4_ENA_MASK 0x0008 #define MADERA_AIF2TX4_ENA_SHIFT 3 -#define MADERA_AIF2TX4_ENA_WIDTH 1 #define MADERA_AIF2TX3_ENA 0x0004 #define MADERA_AIF2TX3_ENA_MASK 0x0004 #define MADERA_AIF2TX3_ENA_SHIFT 2 -#define MADERA_AIF2TX3_ENA_WIDTH 1 #define MADERA_AIF2TX2_ENA 0x0002 #define MADERA_AIF2TX2_ENA_MASK 0x0002 #define MADERA_AIF2TX2_ENA_SHIFT 1 -#define MADERA_AIF2TX2_ENA_WIDTH 1 #define MADERA_AIF2TX1_ENA 0x0001 #define MADERA_AIF2TX1_ENA_MASK 0x0001 #define MADERA_AIF2TX1_ENA_SHIFT 0 -#define MADERA_AIF2TX1_ENA_WIDTH 1 /* (0x055A) AIF2_Rx_Enables */ #define MADERA_AIF2RX8_ENA 0x0080 #define MADERA_AIF2RX8_ENA_MASK 0x0080 #define MADERA_AIF2RX8_ENA_SHIFT 7 -#define MADERA_AIF2RX8_ENA_WIDTH 1 #define MADERA_AIF2RX7_ENA 0x0040 #define MADERA_AIF2RX7_ENA_MASK 0x0040 #define MADERA_AIF2RX7_ENA_SHIFT 6 -#define MADERA_AIF2RX7_ENA_WIDTH 1 #define MADERA_AIF2RX6_ENA 0x0020 #define MADERA_AIF2RX6_ENA_MASK 0x0020 #define MADERA_AIF2RX6_ENA_SHIFT 5 -#define MADERA_AIF2RX6_ENA_WIDTH 1 #define MADERA_AIF2RX5_ENA 0x0010 #define MADERA_AIF2RX5_ENA_MASK 0x0010 #define MADERA_AIF2RX5_ENA_SHIFT 4 -#define MADERA_AIF2RX5_ENA_WIDTH 1 #define MADERA_AIF2RX4_ENA 0x0008 #define MADERA_AIF2RX4_ENA_MASK 0x0008 #define MADERA_AIF2RX4_ENA_SHIFT 3 -#define MADERA_AIF2RX4_ENA_WIDTH 1 #define MADERA_AIF2RX3_ENA 0x0004 #define MADERA_AIF2RX3_ENA_MASK 0x0004 #define MADERA_AIF2RX3_ENA_SHIFT 2 -#define MADERA_AIF2RX3_ENA_WIDTH 1 #define MADERA_AIF2RX2_ENA 0x0002 #define MADERA_AIF2RX2_ENA_MASK 0x0002 #define MADERA_AIF2RX2_ENA_SHIFT 1 -#define MADERA_AIF2RX2_ENA_WIDTH 1 #define MADERA_AIF2RX1_ENA 0x0001 #define MADERA_AIF2RX1_ENA_MASK 0x0001 #define MADERA_AIF2RX1_ENA_SHIFT 0 -#define MADERA_AIF2RX1_ENA_WIDTH 1 /* (0x0599) AIF3_Tx_Enables */ #define MADERA_AIF3TX8_ENA 0x0080 #define MADERA_AIF3TX8_ENA_MASK 0x0080 #define MADERA_AIF3TX8_ENA_SHIFT 7 -#define MADERA_AIF3TX8_ENA_WIDTH 1 #define MADERA_AIF3TX7_ENA 0x0040 #define MADERA_AIF3TX7_ENA_MASK 0x0040 #define MADERA_AIF3TX7_ENA_SHIFT 6 -#define MADERA_AIF3TX7_ENA_WIDTH 1 #define MADERA_AIF3TX6_ENA 0x0020 #define MADERA_AIF3TX6_ENA_MASK 0x0020 #define MADERA_AIF3TX6_ENA_SHIFT 5 -#define MADERA_AIF3TX6_ENA_WIDTH 1 #define MADERA_AIF3TX5_ENA 0x0010 #define MADERA_AIF3TX5_ENA_MASK 0x0010 #define MADERA_AIF3TX5_ENA_SHIFT 4 -#define MADERA_AIF3TX5_ENA_WIDTH 1 #define MADERA_AIF3TX4_ENA 0x0008 #define MADERA_AIF3TX4_ENA_MASK 0x0008 #define MADERA_AIF3TX4_ENA_SHIFT 3 -#define MADERA_AIF3TX4_ENA_WIDTH 1 #define MADERA_AIF3TX3_ENA 0x0004 #define MADERA_AIF3TX3_ENA_MASK 0x0004 #define MADERA_AIF3TX3_ENA_SHIFT 2 -#define MADERA_AIF3TX3_ENA_WIDTH 1 #define MADERA_AIF3TX2_ENA 0x0002 #define MADERA_AIF3TX2_ENA_MASK 0x0002 #define MADERA_AIF3TX2_ENA_SHIFT 1 -#define MADERA_AIF3TX2_ENA_WIDTH 1 #define MADERA_AIF3TX1_ENA 0x0001 #define MADERA_AIF3TX1_ENA_MASK 0x0001 #define MADERA_AIF3TX1_ENA_SHIFT 0 -#define MADERA_AIF3TX1_ENA_WIDTH 1 /* (0x059A) AIF3_Rx_Enables */ #define MADERA_AIF3RX8_ENA 0x0080 #define MADERA_AIF3RX8_ENA_MASK 0x0080 #define MADERA_AIF3RX8_ENA_SHIFT 7 -#define MADERA_AIF3RX8_ENA_WIDTH 1 #define MADERA_AIF3RX7_ENA 0x0040 #define MADERA_AIF3RX7_ENA_MASK 0x0040 #define MADERA_AIF3RX7_ENA_SHIFT 6 -#define MADERA_AIF3RX7_ENA_WIDTH 1 #define MADERA_AIF3RX6_ENA 0x0020 #define MADERA_AIF3RX6_ENA_MASK 0x0020 #define MADERA_AIF3RX6_ENA_SHIFT 5 -#define MADERA_AIF3RX6_ENA_WIDTH 1 #define MADERA_AIF3RX5_ENA 0x0010 #define MADERA_AIF3RX5_ENA_MASK 0x0010 #define MADERA_AIF3RX5_ENA_SHIFT 4 -#define MADERA_AIF3RX5_ENA_WIDTH 1 #define MADERA_AIF3RX4_ENA 0x0008 #define MADERA_AIF3RX4_ENA_MASK 0x0008 #define MADERA_AIF3RX4_ENA_SHIFT 3 -#define MADERA_AIF3RX4_ENA_WIDTH 1 #define MADERA_AIF3RX3_ENA 0x0004 #define MADERA_AIF3RX3_ENA_MASK 0x0004 #define MADERA_AIF3RX3_ENA_SHIFT 2 -#define MADERA_AIF3RX3_ENA_WIDTH 1 #define MADERA_AIF3RX2_ENA 0x0002 #define MADERA_AIF3RX2_ENA_MASK 0x0002 #define MADERA_AIF3RX2_ENA_SHIFT 1 -#define MADERA_AIF3RX2_ENA_WIDTH 1 #define MADERA_AIF3RX1_ENA 0x0001 #define MADERA_AIF3RX1_ENA_MASK 0x0001 #define MADERA_AIF3RX1_ENA_SHIFT 0 -#define MADERA_AIF3RX1_ENA_WIDTH 1 /* (0x05B9) AIF4_Tx_Enables */ #define MADERA_AIF4TX2_ENA 0x0002 #define MADERA_AIF4TX2_ENA_MASK 0x0002 #define MADERA_AIF4TX2_ENA_SHIFT 1 -#define MADERA_AIF4TX2_ENA_WIDTH 1 #define MADERA_AIF4TX1_ENA 0x0001 #define MADERA_AIF4TX1_ENA_MASK 0x0001 #define MADERA_AIF4TX1_ENA_SHIFT 0 -#define MADERA_AIF4TX1_ENA_WIDTH 1 /* (0x05BA) AIF4_Rx_Enables */ #define MADERA_AIF4RX2_ENA 0x0002 #define MADERA_AIF4RX2_ENA_MASK 0x0002 #define MADERA_AIF4RX2_ENA_SHIFT 1 -#define MADERA_AIF4RX2_ENA_WIDTH 1 #define MADERA_AIF4RX1_ENA 0x0001 #define MADERA_AIF4RX1_ENA_MASK 0x0001 #define MADERA_AIF4RX1_ENA_SHIFT 0 -#define MADERA_AIF4RX1_ENA_WIDTH 1 /* (0x05C2) SPD1_TX_Control */ #define MADERA_SPD1_VAL2 0x2000 #define MADERA_SPD1_VAL2_MASK 0x2000 #define MADERA_SPD1_VAL2_SHIFT 13 -#define MADERA_SPD1_VAL2_WIDTH 1 #define MADERA_SPD1_VAL1 0x1000 #define MADERA_SPD1_VAL1_MASK 0x1000 #define MADERA_SPD1_VAL1_SHIFT 12 -#define MADERA_SPD1_VAL1_WIDTH 1 #define MADERA_SPD1_RATE_MASK 0x00F0 #define MADERA_SPD1_RATE_SHIFT 4 -#define MADERA_SPD1_RATE_WIDTH 4 #define MADERA_SPD1_ENA 0x0001 #define MADERA_SPD1_ENA_MASK 0x0001 #define MADERA_SPD1_ENA_SHIFT 0 -#define MADERA_SPD1_ENA_WIDTH 1 /* (0x05F5) SLIMbus_RX_Channel_Enable */ #define MADERA_SLIMRX8_ENA 0x0080 #define MADERA_SLIMRX8_ENA_MASK 0x0080 #define MADERA_SLIMRX8_ENA_SHIFT 7 -#define MADERA_SLIMRX8_ENA_WIDTH 1 #define MADERA_SLIMRX7_ENA 0x0040 #define MADERA_SLIMRX7_ENA_MASK 0x0040 #define MADERA_SLIMRX7_ENA_SHIFT 6 -#define MADERA_SLIMRX7_ENA_WIDTH 1 #define MADERA_SLIMRX6_ENA 0x0020 #define MADERA_SLIMRX6_ENA_MASK 0x0020 #define MADERA_SLIMRX6_ENA_SHIFT 5 -#define MADERA_SLIMRX6_ENA_WIDTH 1 #define MADERA_SLIMRX5_ENA 0x0010 #define MADERA_SLIMRX5_ENA_MASK 0x0010 #define MADERA_SLIMRX5_ENA_SHIFT 4 -#define MADERA_SLIMRX5_ENA_WIDTH 1 #define MADERA_SLIMRX4_ENA 0x0008 #define MADERA_SLIMRX4_ENA_MASK 0x0008 #define MADERA_SLIMRX4_ENA_SHIFT 3 -#define MADERA_SLIMRX4_ENA_WIDTH 1 #define MADERA_SLIMRX3_ENA 0x0004 #define MADERA_SLIMRX3_ENA_MASK 0x0004 #define MADERA_SLIMRX3_ENA_SHIFT 2 -#define MADERA_SLIMRX3_ENA_WIDTH 1 #define MADERA_SLIMRX2_ENA 0x0002 #define MADERA_SLIMRX2_ENA_MASK 0x0002 #define MADERA_SLIMRX2_ENA_SHIFT 1 -#define MADERA_SLIMRX2_ENA_WIDTH 1 #define MADERA_SLIMRX1_ENA 0x0001 #define MADERA_SLIMRX1_ENA_MASK 0x0001 #define MADERA_SLIMRX1_ENA_SHIFT 0 -#define MADERA_SLIMRX1_ENA_WIDTH 1 /* (0x05F6) SLIMbus_TX_Channel_Enable */ #define MADERA_SLIMTX8_ENA 0x0080 #define MADERA_SLIMTX8_ENA_MASK 0x0080 #define MADERA_SLIMTX8_ENA_SHIFT 7 -#define MADERA_SLIMTX8_ENA_WIDTH 1 #define MADERA_SLIMTX7_ENA 0x0040 #define MADERA_SLIMTX7_ENA_MASK 0x0040 #define MADERA_SLIMTX7_ENA_SHIFT 6 -#define MADERA_SLIMTX7_ENA_WIDTH 1 #define MADERA_SLIMTX6_ENA 0x0020 #define MADERA_SLIMTX6_ENA_MASK 0x0020 #define MADERA_SLIMTX6_ENA_SHIFT 5 -#define MADERA_SLIMTX6_ENA_WIDTH 1 #define MADERA_SLIMTX5_ENA 0x0010 #define MADERA_SLIMTX5_ENA_MASK 0x0010 #define MADERA_SLIMTX5_ENA_SHIFT 4 -#define MADERA_SLIMTX5_ENA_WIDTH 1 #define MADERA_SLIMTX4_ENA 0x0008 #define MADERA_SLIMTX4_ENA_MASK 0x0008 #define MADERA_SLIMTX4_ENA_SHIFT 3 -#define MADERA_SLIMTX4_ENA_WIDTH 1 #define MADERA_SLIMTX3_ENA 0x0004 #define MADERA_SLIMTX3_ENA_MASK 0x0004 #define MADERA_SLIMTX3_ENA_SHIFT 2 -#define MADERA_SLIMTX3_ENA_WIDTH 1 #define MADERA_SLIMTX2_ENA 0x0002 #define MADERA_SLIMTX2_ENA_MASK 0x0002 #define MADERA_SLIMTX2_ENA_SHIFT 1 -#define MADERA_SLIMTX2_ENA_WIDTH 1 #define MADERA_SLIMTX1_ENA 0x0001 #define MADERA_SLIMTX1_ENA_MASK 0x0001 #define MADERA_SLIMTX1_ENA_SHIFT 0 -#define MADERA_SLIMTX1_ENA_WIDTH 1 /* (0x0E10) EQ1_1 */ #define MADERA_EQ1_B1_GAIN_MASK 0xF800 #define MADERA_EQ1_B1_GAIN_SHIFT 11 -#define MADERA_EQ1_B1_GAIN_WIDTH 5 #define MADERA_EQ1_B2_GAIN_MASK 0x07C0 #define MADERA_EQ1_B2_GAIN_SHIFT 6 -#define MADERA_EQ1_B2_GAIN_WIDTH 5 #define MADERA_EQ1_B3_GAIN_MASK 0x003E #define MADERA_EQ1_B3_GAIN_SHIFT 1 -#define MADERA_EQ1_B3_GAIN_WIDTH 5 #define MADERA_EQ1_ENA 0x0001 #define MADERA_EQ1_ENA_MASK 0x0001 #define MADERA_EQ1_ENA_SHIFT 0 -#define MADERA_EQ1_ENA_WIDTH 1 /* (0x0E11) EQ1_2 */ #define MADERA_EQ1_B4_GAIN_MASK 0xF800 #define MADERA_EQ1_B4_GAIN_SHIFT 11 -#define MADERA_EQ1_B4_GAIN_WIDTH 5 #define MADERA_EQ1_B5_GAIN_MASK 0x07C0 #define MADERA_EQ1_B5_GAIN_SHIFT 6 -#define MADERA_EQ1_B5_GAIN_WIDTH 5 #define MADERA_EQ1_B1_MODE 0x0001 #define MADERA_EQ1_B1_MODE_MASK 0x0001 #define MADERA_EQ1_B1_MODE_SHIFT 0 -#define MADERA_EQ1_B1_MODE_WIDTH 1 /* (0x0E26) EQ2_1 */ #define MADERA_EQ2_B1_GAIN_MASK 0xF800 #define MADERA_EQ2_B1_GAIN_SHIFT 11 -#define MADERA_EQ2_B1_GAIN_WIDTH 5 #define MADERA_EQ2_B2_GAIN_MASK 0x07C0 #define MADERA_EQ2_B2_GAIN_SHIFT 6 -#define MADERA_EQ2_B2_GAIN_WIDTH 5 #define MADERA_EQ2_B3_GAIN_MASK 0x003E #define MADERA_EQ2_B3_GAIN_SHIFT 1 -#define MADERA_EQ2_B3_GAIN_WIDTH 5 #define MADERA_EQ2_ENA 0x0001 #define MADERA_EQ2_ENA_MASK 0x0001 #define MADERA_EQ2_ENA_SHIFT 0 -#define MADERA_EQ2_ENA_WIDTH 1 /* (0x0E27) EQ2_2 */ #define MADERA_EQ2_B4_GAIN_MASK 0xF800 #define MADERA_EQ2_B4_GAIN_SHIFT 11 -#define MADERA_EQ2_B4_GAIN_WIDTH 5 #define MADERA_EQ2_B5_GAIN_MASK 0x07C0 #define MADERA_EQ2_B5_GAIN_SHIFT 6 -#define MADERA_EQ2_B5_GAIN_WIDTH 5 #define MADERA_EQ2_B1_MODE 0x0001 #define MADERA_EQ2_B1_MODE_MASK 0x0001 #define MADERA_EQ2_B1_MODE_SHIFT 0 -#define MADERA_EQ2_B1_MODE_WIDTH 1 /* (0x0E3C) EQ3_1 */ #define MADERA_EQ3_B1_GAIN_MASK 0xF800 #define MADERA_EQ3_B1_GAIN_SHIFT 11 -#define MADERA_EQ3_B1_GAIN_WIDTH 5 #define MADERA_EQ3_B2_GAIN_MASK 0x07C0 #define MADERA_EQ3_B2_GAIN_SHIFT 6 -#define MADERA_EQ3_B2_GAIN_WIDTH 5 #define MADERA_EQ3_B3_GAIN_MASK 0x003E #define MADERA_EQ3_B3_GAIN_SHIFT 1 -#define MADERA_EQ3_B3_GAIN_WIDTH 5 #define MADERA_EQ3_ENA 0x0001 #define MADERA_EQ3_ENA_MASK 0x0001 #define MADERA_EQ3_ENA_SHIFT 0 -#define MADERA_EQ3_ENA_WIDTH 1 /* (0x0E3D) EQ3_2 */ #define MADERA_EQ3_B4_GAIN_MASK 0xF800 #define MADERA_EQ3_B4_GAIN_SHIFT 11 -#define MADERA_EQ3_B4_GAIN_WIDTH 5 #define MADERA_EQ3_B5_GAIN_MASK 0x07C0 #define MADERA_EQ3_B5_GAIN_SHIFT 6 -#define MADERA_EQ3_B5_GAIN_WIDTH 5 #define MADERA_EQ3_B1_MODE 0x0001 #define MADERA_EQ3_B1_MODE_MASK 0x0001 #define MADERA_EQ3_B1_MODE_SHIFT 0 -#define MADERA_EQ3_B1_MODE_WIDTH 1 /* (0x0E52) EQ4_1 */ #define MADERA_EQ4_B1_GAIN_MASK 0xF800 #define MADERA_EQ4_B1_GAIN_SHIFT 11 -#define MADERA_EQ4_B1_GAIN_WIDTH 5 #define MADERA_EQ4_B2_GAIN_MASK 0x07C0 #define MADERA_EQ4_B2_GAIN_SHIFT 6 -#define MADERA_EQ4_B2_GAIN_WIDTH 5 #define MADERA_EQ4_B3_GAIN_MASK 0x003E #define MADERA_EQ4_B3_GAIN_SHIFT 1 -#define MADERA_EQ4_B3_GAIN_WIDTH 5 #define MADERA_EQ4_ENA 0x0001 #define MADERA_EQ4_ENA_MASK 0x0001 #define MADERA_EQ4_ENA_SHIFT 0 -#define MADERA_EQ4_ENA_WIDTH 1 /* (0x0E53) EQ4_2 */ #define MADERA_EQ4_B4_GAIN_MASK 0xF800 #define MADERA_EQ4_B4_GAIN_SHIFT 11 -#define MADERA_EQ4_B4_GAIN_WIDTH 5 #define MADERA_EQ4_B5_GAIN_MASK 0x07C0 #define MADERA_EQ4_B5_GAIN_SHIFT 6 -#define MADERA_EQ4_B5_GAIN_WIDTH 5 #define MADERA_EQ4_B1_MODE 0x0001 #define MADERA_EQ4_B1_MODE_MASK 0x0001 #define MADERA_EQ4_B1_MODE_SHIFT 0 -#define MADERA_EQ4_B1_MODE_WIDTH 1 /* (0x0E80) DRC1_ctrl1 */ #define MADERA_DRC1L_ENA 0x0002 #define MADERA_DRC1L_ENA_MASK 0x0002 #define MADERA_DRC1L_ENA_SHIFT 1 -#define MADERA_DRC1L_ENA_WIDTH 1 #define MADERA_DRC1R_ENA 0x0001 #define MADERA_DRC1R_ENA_MASK 0x0001 #define MADERA_DRC1R_ENA_SHIFT 0 -#define MADERA_DRC1R_ENA_WIDTH 1 /* (0x0E88) DRC2_ctrl1 */ #define MADERA_DRC2L_ENA 0x0002 #define MADERA_DRC2L_ENA_MASK 0x0002 #define MADERA_DRC2L_ENA_SHIFT 1 -#define MADERA_DRC2L_ENA_WIDTH 1 #define MADERA_DRC2R_ENA 0x0001 #define MADERA_DRC2R_ENA_MASK 0x0001 #define MADERA_DRC2R_ENA_SHIFT 0 -#define MADERA_DRC2R_ENA_WIDTH 1 /* (0x0EC0) HPLPF1_1 */ #define MADERA_LHPF1_MODE 0x0002 #define MADERA_LHPF1_MODE_MASK 0x0002 #define MADERA_LHPF1_MODE_SHIFT 1 -#define MADERA_LHPF1_MODE_WIDTH 1 #define MADERA_LHPF1_ENA 0x0001 #define MADERA_LHPF1_ENA_MASK 0x0001 #define MADERA_LHPF1_ENA_SHIFT 0 -#define MADERA_LHPF1_ENA_WIDTH 1 /* (0x0EC1) HPLPF1_2 */ #define MADERA_LHPF1_COEFF_MASK 0xFFFF #define MADERA_LHPF1_COEFF_SHIFT 0 -#define MADERA_LHPF1_COEFF_WIDTH 16 /* (0x0EC4) HPLPF2_1 */ #define MADERA_LHPF2_MODE 0x0002 #define MADERA_LHPF2_MODE_MASK 0x0002 #define MADERA_LHPF2_MODE_SHIFT 1 -#define MADERA_LHPF2_MODE_WIDTH 1 #define MADERA_LHPF2_ENA 0x0001 #define MADERA_LHPF2_ENA_MASK 0x0001 #define MADERA_LHPF2_ENA_SHIFT 0 -#define MADERA_LHPF2_ENA_WIDTH 1 /* (0x0EC5) HPLPF2_2 */ #define MADERA_LHPF2_COEFF_MASK 0xFFFF #define MADERA_LHPF2_COEFF_SHIFT 0 -#define MADERA_LHPF2_COEFF_WIDTH 16 /* (0x0EC8) HPLPF3_1 */ #define MADERA_LHPF3_MODE 0x0002 #define MADERA_LHPF3_MODE_MASK 0x0002 #define MADERA_LHPF3_MODE_SHIFT 1 -#define MADERA_LHPF3_MODE_WIDTH 1 #define MADERA_LHPF3_ENA 0x0001 #define MADERA_LHPF3_ENA_MASK 0x0001 #define MADERA_LHPF3_ENA_SHIFT 0 -#define MADERA_LHPF3_ENA_WIDTH 1 /* (0x0EC9) HPLPF3_2 */ #define MADERA_LHPF3_COEFF_MASK 0xFFFF #define MADERA_LHPF3_COEFF_SHIFT 0 -#define MADERA_LHPF3_COEFF_WIDTH 16 /* (0x0ECC) HPLPF4_1 */ #define MADERA_LHPF4_MODE 0x0002 #define MADERA_LHPF4_MODE_MASK 0x0002 #define MADERA_LHPF4_MODE_SHIFT 1 -#define MADERA_LHPF4_MODE_WIDTH 1 #define MADERA_LHPF4_ENA 0x0001 #define MADERA_LHPF4_ENA_MASK 0x0001 #define MADERA_LHPF4_ENA_SHIFT 0 -#define MADERA_LHPF4_ENA_WIDTH 1 /* (0x0ECD) HPLPF4_2 */ #define MADERA_LHPF4_COEFF_MASK 0xFFFF #define MADERA_LHPF4_COEFF_SHIFT 0 -#define MADERA_LHPF4_COEFF_WIDTH 16 /* (0x0ED0) ASRC2_ENABLE */ #define MADERA_ASRC2_IN2L_ENA 0x0008 #define MADERA_ASRC2_IN2L_ENA_MASK 0x0008 #define MADERA_ASRC2_IN2L_ENA_SHIFT 3 -#define MADERA_ASRC2_IN2L_ENA_WIDTH 1 #define MADERA_ASRC2_IN2R_ENA 0x0004 #define MADERA_ASRC2_IN2R_ENA_MASK 0x0004 #define MADERA_ASRC2_IN2R_ENA_SHIFT 2 -#define MADERA_ASRC2_IN2R_ENA_WIDTH 1 #define MADERA_ASRC2_IN1L_ENA 0x0002 #define MADERA_ASRC2_IN1L_ENA_MASK 0x0002 #define MADERA_ASRC2_IN1L_ENA_SHIFT 1 -#define MADERA_ASRC2_IN1L_ENA_WIDTH 1 #define MADERA_ASRC2_IN1R_ENA 0x0001 #define MADERA_ASRC2_IN1R_ENA_MASK 0x0001 #define MADERA_ASRC2_IN1R_ENA_SHIFT 0 -#define MADERA_ASRC2_IN1R_ENA_WIDTH 1 /* (0x0ED2) ASRC2_RATE1 */ #define MADERA_ASRC2_RATE1_MASK 0xF800 #define MADERA_ASRC2_RATE1_SHIFT 11 -#define MADERA_ASRC2_RATE1_WIDTH 5 /* (0x0ED3) ASRC2_RATE2 */ #define MADERA_ASRC2_RATE2_MASK 0xF800 #define MADERA_ASRC2_RATE2_SHIFT 11 -#define MADERA_ASRC2_RATE2_WIDTH 5 /* (0x0EE0) ASRC1_ENABLE */ #define MADERA_ASRC1_IN2L_ENA 0x0008 #define MADERA_ASRC1_IN2L_ENA_MASK 0x0008 #define MADERA_ASRC1_IN2L_ENA_SHIFT 3 -#define MADERA_ASRC1_IN2L_ENA_WIDTH 1 #define MADERA_ASRC1_IN2R_ENA 0x0004 #define MADERA_ASRC1_IN2R_ENA_MASK 0x0004 #define MADERA_ASRC1_IN2R_ENA_SHIFT 2 -#define MADERA_ASRC1_IN2R_ENA_WIDTH 1 #define MADERA_ASRC1_IN1L_ENA 0x0002 #define MADERA_ASRC1_IN1L_ENA_MASK 0x0002 #define MADERA_ASRC1_IN1L_ENA_SHIFT 1 -#define MADERA_ASRC1_IN1L_ENA_WIDTH 1 #define MADERA_ASRC1_IN1R_ENA 0x0001 #define MADERA_ASRC1_IN1R_ENA_MASK 0x0001 #define MADERA_ASRC1_IN1R_ENA_SHIFT 0 -#define MADERA_ASRC1_IN1R_ENA_WIDTH 1 /* (0x0EE2) ASRC1_RATE1 */ #define MADERA_ASRC1_RATE1_MASK 0xF800 #define MADERA_ASRC1_RATE1_SHIFT 11 -#define MADERA_ASRC1_RATE1_WIDTH 5 /* (0x0EE3) ASRC1_RATE2 */ #define MADERA_ASRC1_RATE2_MASK 0xF800 #define MADERA_ASRC1_RATE2_SHIFT 11 -#define MADERA_ASRC1_RATE2_WIDTH 5 /* (0x0EF0) - ISRC1 CTRL 1 */ #define MADERA_ISRC1_FSH_MASK 0xF800 #define MADERA_ISRC1_FSH_SHIFT 11 -#define MADERA_ISRC1_FSH_WIDTH 5 #define MADERA_ISRC1_CLK_SEL_MASK 0x0700 #define MADERA_ISRC1_CLK_SEL_SHIFT 8 -#define MADERA_ISRC1_CLK_SEL_WIDTH 3 /* (0x0EF1) ISRC1_CTRL_2 */ #define MADERA_ISRC1_FSL_MASK 0xF800 #define MADERA_ISRC1_FSL_SHIFT 11 -#define MADERA_ISRC1_FSL_WIDTH 5 /* (0x0EF2) ISRC1_CTRL_3 */ #define MADERA_ISRC1_INT1_ENA 0x8000 #define MADERA_ISRC1_INT1_ENA_MASK 0x8000 #define MADERA_ISRC1_INT1_ENA_SHIFT 15 -#define MADERA_ISRC1_INT1_ENA_WIDTH 1 #define MADERA_ISRC1_INT2_ENA 0x4000 #define MADERA_ISRC1_INT2_ENA_MASK 0x4000 #define MADERA_ISRC1_INT2_ENA_SHIFT 14 -#define MADERA_ISRC1_INT2_ENA_WIDTH 1 #define MADERA_ISRC1_INT3_ENA 0x2000 #define MADERA_ISRC1_INT3_ENA_MASK 0x2000 #define MADERA_ISRC1_INT3_ENA_SHIFT 13 -#define MADERA_ISRC1_INT3_ENA_WIDTH 1 #define MADERA_ISRC1_INT4_ENA 0x1000 #define MADERA_ISRC1_INT4_ENA_MASK 0x1000 #define MADERA_ISRC1_INT4_ENA_SHIFT 12 -#define MADERA_ISRC1_INT4_ENA_WIDTH 1 #define MADERA_ISRC1_DEC1_ENA 0x0200 #define MADERA_ISRC1_DEC1_ENA_MASK 0x0200 #define MADERA_ISRC1_DEC1_ENA_SHIFT 9 -#define MADERA_ISRC1_DEC1_ENA_WIDTH 1 #define MADERA_ISRC1_DEC2_ENA 0x0100 #define MADERA_ISRC1_DEC2_ENA_MASK 0x0100 #define MADERA_ISRC1_DEC2_ENA_SHIFT 8 -#define MADERA_ISRC1_DEC2_ENA_WIDTH 1 #define MADERA_ISRC1_DEC3_ENA 0x0080 #define MADERA_ISRC1_DEC3_ENA_MASK 0x0080 #define MADERA_ISRC1_DEC3_ENA_SHIFT 7 -#define MADERA_ISRC1_DEC3_ENA_WIDTH 1 #define MADERA_ISRC1_DEC4_ENA 0x0040 #define MADERA_ISRC1_DEC4_ENA_MASK 0x0040 #define MADERA_ISRC1_DEC4_ENA_SHIFT 6 -#define MADERA_ISRC1_DEC4_ENA_WIDTH 1 #define MADERA_ISRC1_NOTCH_ENA 0x0001 #define MADERA_ISRC1_NOTCH_ENA_MASK 0x0001 #define MADERA_ISRC1_NOTCH_ENA_SHIFT 0 -#define MADERA_ISRC1_NOTCH_ENA_WIDTH 1 /* (0x0EF3) ISRC2_CTRL_1 */ #define MADERA_ISRC2_FSH_MASK 0xF800 #define MADERA_ISRC2_FSH_SHIFT 11 -#define MADERA_ISRC2_FSH_WIDTH 5 #define MADERA_ISRC2_CLK_SEL_MASK 0x0700 #define MADERA_ISRC2_CLK_SEL_SHIFT 8 -#define MADERA_ISRC2_CLK_SEL_WIDTH 3 /* (0x0EF4) ISRC2_CTRL_2 */ #define MADERA_ISRC2_FSL_MASK 0xF800 #define MADERA_ISRC2_FSL_SHIFT 11 -#define MADERA_ISRC2_FSL_WIDTH 5 /* (0x0EF5) ISRC2_CTRL_3 */ #define MADERA_ISRC2_INT1_ENA 0x8000 #define MADERA_ISRC2_INT1_ENA_MASK 0x8000 #define MADERA_ISRC2_INT1_ENA_SHIFT 15 -#define MADERA_ISRC2_INT1_ENA_WIDTH 1 #define MADERA_ISRC2_INT2_ENA 0x4000 #define MADERA_ISRC2_INT2_ENA_MASK 0x4000 #define MADERA_ISRC2_INT2_ENA_SHIFT 14 -#define MADERA_ISRC2_INT2_ENA_WIDTH 1 #define MADERA_ISRC2_INT3_ENA 0x2000 #define MADERA_ISRC2_INT3_ENA_MASK 0x2000 #define MADERA_ISRC2_INT3_ENA_SHIFT 13 -#define MADERA_ISRC2_INT3_ENA_WIDTH 1 #define MADERA_ISRC2_INT4_ENA 0x1000 #define MADERA_ISRC2_INT4_ENA_MASK 0x1000 #define MADERA_ISRC2_INT4_ENA_SHIFT 12 -#define MADERA_ISRC2_INT4_ENA_WIDTH 1 #define MADERA_ISRC2_DEC1_ENA 0x0200 #define MADERA_ISRC2_DEC1_ENA_MASK 0x0200 #define MADERA_ISRC2_DEC1_ENA_SHIFT 9 -#define MADERA_ISRC2_DEC1_ENA_WIDTH 1 #define MADERA_ISRC2_DEC2_ENA 0x0100 #define MADERA_ISRC2_DEC2_ENA_MASK 0x0100 #define MADERA_ISRC2_DEC2_ENA_SHIFT 8 -#define MADERA_ISRC2_DEC2_ENA_WIDTH 1 #define MADERA_ISRC2_DEC3_ENA 0x0080 #define MADERA_ISRC2_DEC3_ENA_MASK 0x0080 #define MADERA_ISRC2_DEC3_ENA_SHIFT 7 -#define MADERA_ISRC2_DEC3_ENA_WIDTH 1 #define MADERA_ISRC2_DEC4_ENA 0x0040 #define MADERA_ISRC2_DEC4_ENA_MASK 0x0040 #define MADERA_ISRC2_DEC4_ENA_SHIFT 6 -#define MADERA_ISRC2_DEC4_ENA_WIDTH 1 #define MADERA_ISRC2_NOTCH_ENA 0x0001 #define MADERA_ISRC2_NOTCH_ENA_MASK 0x0001 #define MADERA_ISRC2_NOTCH_ENA_SHIFT 0 -#define MADERA_ISRC2_NOTCH_ENA_WIDTH 1 /* (0x0EF6) ISRC3_CTRL_1 */ #define MADERA_ISRC3_FSH_MASK 0xF800 #define MADERA_ISRC3_FSH_SHIFT 11 -#define MADERA_ISRC3_FSH_WIDTH 5 #define MADERA_ISRC3_CLK_SEL_MASK 0x0700 #define MADERA_ISRC3_CLK_SEL_SHIFT 8 -#define MADERA_ISRC3_CLK_SEL_WIDTH 3 /* (0x0EF7) ISRC3_CTRL_2 */ #define MADERA_ISRC3_FSL_MASK 0xF800 #define MADERA_ISRC3_FSL_SHIFT 11 -#define MADERA_ISRC3_FSL_WIDTH 5 /* (0x0EF8) ISRC3_CTRL_3 */ #define MADERA_ISRC3_INT1_ENA 0x8000 #define MADERA_ISRC3_INT1_ENA_MASK 0x8000 #define MADERA_ISRC3_INT1_ENA_SHIFT 15 -#define MADERA_ISRC3_INT1_ENA_WIDTH 1 #define MADERA_ISRC3_INT2_ENA 0x4000 #define MADERA_ISRC3_INT2_ENA_MASK 0x4000 #define MADERA_ISRC3_INT2_ENA_SHIFT 14 -#define MADERA_ISRC3_INT2_ENA_WIDTH 1 #define MADERA_ISRC3_INT3_ENA 0x2000 #define MADERA_ISRC3_INT3_ENA_MASK 0x2000 #define MADERA_ISRC3_INT3_ENA_SHIFT 13 -#define MADERA_ISRC3_INT3_ENA_WIDTH 1 #define MADERA_ISRC3_INT4_ENA 0x1000 #define MADERA_ISRC3_INT4_ENA_MASK 0x1000 #define MADERA_ISRC3_INT4_ENA_SHIFT 12 -#define MADERA_ISRC3_INT4_ENA_WIDTH 1 #define MADERA_ISRC3_DEC1_ENA 0x0200 #define MADERA_ISRC3_DEC1_ENA_MASK 0x0200 #define MADERA_ISRC3_DEC1_ENA_SHIFT 9 -#define MADERA_ISRC3_DEC1_ENA_WIDTH 1 #define MADERA_ISRC3_DEC2_ENA 0x0100 #define MADERA_ISRC3_DEC2_ENA_MASK 0x0100 #define MADERA_ISRC3_DEC2_ENA_SHIFT 8 -#define MADERA_ISRC3_DEC2_ENA_WIDTH 1 #define MADERA_ISRC3_DEC3_ENA 0x0080 #define MADERA_ISRC3_DEC3_ENA_MASK 0x0080 #define MADERA_ISRC3_DEC3_ENA_SHIFT 7 -#define MADERA_ISRC3_DEC3_ENA_WIDTH 1 #define MADERA_ISRC3_DEC4_ENA 0x0040 #define MADERA_ISRC3_DEC4_ENA_MASK 0x0040 #define MADERA_ISRC3_DEC4_ENA_SHIFT 6 -#define MADERA_ISRC3_DEC4_ENA_WIDTH 1 #define MADERA_ISRC3_NOTCH_ENA 0x0001 #define MADERA_ISRC3_NOTCH_ENA_MASK 0x0001 #define MADERA_ISRC3_NOTCH_ENA_SHIFT 0 -#define MADERA_ISRC3_NOTCH_ENA_WIDTH 1 /* (0x0EF9) ISRC4_CTRL_1 */ #define MADERA_ISRC4_FSH_MASK 0xF800 #define MADERA_ISRC4_FSH_SHIFT 11 -#define MADERA_ISRC4_FSH_WIDTH 5 #define MADERA_ISRC4_CLK_SEL_MASK 0x0700 #define MADERA_ISRC4_CLK_SEL_SHIFT 8 -#define MADERA_ISRC4_CLK_SEL_WIDTH 3 /* (0x0EFA) ISRC4_CTRL_2 */ #define MADERA_ISRC4_FSL_MASK 0xF800 #define MADERA_ISRC4_FSL_SHIFT 11 -#define MADERA_ISRC4_FSL_WIDTH 5 /* (0x0EFB) ISRC4_CTRL_3 */ #define MADERA_ISRC4_INT1_ENA 0x8000 #define MADERA_ISRC4_INT1_ENA_MASK 0x8000 #define MADERA_ISRC4_INT1_ENA_SHIFT 15 -#define MADERA_ISRC4_INT1_ENA_WIDTH 1 #define MADERA_ISRC4_INT2_ENA 0x4000 #define MADERA_ISRC4_INT2_ENA_MASK 0x4000 #define MADERA_ISRC4_INT2_ENA_SHIFT 14 -#define MADERA_ISRC4_INT2_ENA_WIDTH 1 #define MADERA_ISRC4_INT3_ENA 0x2000 #define MADERA_ISRC4_INT3_ENA_MASK 0x2000 #define MADERA_ISRC4_INT3_ENA_SHIFT 13 -#define MADERA_ISRC4_INT3_ENA_WIDTH 1 #define MADERA_ISRC4_INT4_ENA 0x1000 #define MADERA_ISRC4_INT4_ENA_MASK 0x1000 #define MADERA_ISRC4_INT4_ENA_SHIFT 12 -#define MADERA_ISRC4_INT4_ENA_WIDTH 1 #define MADERA_ISRC4_DEC1_ENA 0x0200 #define MADERA_ISRC4_DEC1_ENA_MASK 0x0200 #define MADERA_ISRC4_DEC1_ENA_SHIFT 9 -#define MADERA_ISRC4_DEC1_ENA_WIDTH 1 #define MADERA_ISRC4_DEC2_ENA 0x0100 #define MADERA_ISRC4_DEC2_ENA_MASK 0x0100 #define MADERA_ISRC4_DEC2_ENA_SHIFT 8 -#define MADERA_ISRC4_DEC2_ENA_WIDTH 1 #define MADERA_ISRC4_DEC3_ENA 0x0080 #define MADERA_ISRC4_DEC3_ENA_MASK 0x0080 #define MADERA_ISRC4_DEC3_ENA_SHIFT 7 -#define MADERA_ISRC4_DEC3_ENA_WIDTH 1 #define MADERA_ISRC4_DEC4_ENA 0x0040 #define MADERA_ISRC4_DEC4_ENA_MASK 0x0040 #define MADERA_ISRC4_DEC4_ENA_SHIFT 6 -#define MADERA_ISRC4_DEC4_ENA_WIDTH 1 #define MADERA_ISRC4_NOTCH_ENA 0x0001 #define MADERA_ISRC4_NOTCH_ENA_MASK 0x0001 #define MADERA_ISRC4_NOTCH_ENA_SHIFT 0 -#define MADERA_ISRC4_NOTCH_ENA_WIDTH 1 /* (0x0F00) Clock_Control */ #define MADERA_EXT_NG_SEL_CLR 0x0080 #define MADERA_EXT_NG_SEL_CLR_MASK 0x0080 #define MADERA_EXT_NG_SEL_CLR_SHIFT 7 -#define MADERA_EXT_NG_SEL_CLR_WIDTH 1 #define MADERA_EXT_NG_SEL_SET 0x0040 #define MADERA_EXT_NG_SEL_SET_MASK 0x0040 #define MADERA_EXT_NG_SEL_SET_SHIFT 6 -#define MADERA_EXT_NG_SEL_SET_WIDTH 1 #define MADERA_CLK_R_ENA_CLR 0x0020 #define MADERA_CLK_R_ENA_CLR_MASK 0x0020 #define MADERA_CLK_R_ENA_CLR_SHIFT 5 -#define MADERA_CLK_R_ENA_CLR_WIDTH 1 #define MADERA_CLK_R_ENA_SET 0x0010 #define MADERA_CLK_R_ENA_SET_MASK 0x0010 #define MADERA_CLK_R_ENA_SET_SHIFT 4 -#define MADERA_CLK_R_ENA_SET_WIDTH 1 #define MADERA_CLK_NG_ENA_CLR 0x0008 #define MADERA_CLK_NG_ENA_CLR_MASK 0x0008 #define MADERA_CLK_NG_ENA_CLR_SHIFT 3 -#define MADERA_CLK_NG_ENA_CLR_WIDTH 1 #define MADERA_CLK_NG_ENA_SET 0x0004 #define MADERA_CLK_NG_ENA_SET_MASK 0x0004 #define MADERA_CLK_NG_ENA_SET_SHIFT 2 -#define MADERA_CLK_NG_ENA_SET_WIDTH 1 #define MADERA_CLK_L_ENA_CLR 0x0002 #define MADERA_CLK_L_ENA_CLR_MASK 0x0002 #define MADERA_CLK_L_ENA_CLR_SHIFT 1 -#define MADERA_CLK_L_ENA_CLR_WIDTH 1 #define MADERA_CLK_L_ENA_SET 0x0001 #define MADERA_CLK_L_ENA_SET_MASK 0x0001 #define MADERA_CLK_L_ENA_SET_SHIFT 0 -#define MADERA_CLK_L_ENA_SET_WIDTH 1 /* (0x0F01) ANC_SRC */ #define MADERA_IN_RXANCR_SEL_MASK 0x0070 #define MADERA_IN_RXANCR_SEL_SHIFT 4 -#define MADERA_IN_RXANCR_SEL_WIDTH 3 #define MADERA_IN_RXANCL_SEL_MASK 0x0007 #define MADERA_IN_RXANCL_SEL_SHIFT 0 -#define MADERA_IN_RXANCL_SEL_WIDTH 3 /* (0x0F17) FCL_ADC_reformatter_control */ #define MADERA_FCL_MIC_MODE_SEL 0x000C #define MADERA_FCL_MIC_MODE_SEL_SHIFT 2 -#define MADERA_FCL_MIC_MODE_SEL_WIDTH 2 /* (0x0F73) FCR_ADC_reformatter_control */ #define MADERA_FCR_MIC_MODE_SEL 0x000C #define MADERA_FCR_MIC_MODE_SEL_SHIFT 2 -#define MADERA_FCR_MIC_MODE_SEL_WIDTH 2 /* (0x10C0) AUXPDM1_CTRL_0 */ #define MADERA_AUXPDM1_SRC_MASK 0x0F00 #define MADERA_AUXPDM1_SRC_SHIFT 8 -#define MADERA_AUXPDM1_SRC_WIDTH 4 #define MADERA_AUXPDM1_TXEDGE_MASK 0x0010 #define MADERA_AUXPDM1_TXEDGE_SHIFT 4 -#define MADERA_AUXPDM1_TXEDGE_WIDTH 1 #define MADERA_AUXPDM1_MSTR_MASK 0x0008 #define MADERA_AUXPDM1_MSTR_SHIFT 3 -#define MADERA_AUXPDM1_MSTR_WIDTH 1 #define MADERA_AUXPDM1_ENABLE_MASK 0x0001 #define MADERA_AUXPDM1_ENABLE_SHIFT 0 -#define MADERA_AUXPDM1_ENABLE_WIDTH 1 /* (0x10C1) AUXPDM1_CTRL_1 */ #define MADERA_AUXPDM1_CLK_FREQ_MASK 0xC000 #define MADERA_AUXPDM1_CLK_FREQ_SHIFT 14 -#define MADERA_AUXPDM1_CLK_FREQ_WIDTH 2 /* (0x1480) DFC1_CTRL_W0 */ #define MADERA_DFC1_RATE_MASK 0x007C #define MADERA_DFC1_RATE_SHIFT 2 -#define MADERA_DFC1_RATE_WIDTH 5 #define MADERA_DFC1_DITH_ENA 0x0002 #define MADERA_DFC1_DITH_ENA_MASK 0x0002 #define MADERA_DFC1_DITH_ENA_SHIFT 1 -#define MADERA_DFC1_DITH_ENA_WIDTH 1 #define MADERA_DFC1_ENA 0x0001 #define MADERA_DFC1_ENA_MASK 0x0001 #define MADERA_DFC1_ENA_SHIFT 0 -#define MADERA_DFC1_ENA_WIDTH 1 /* (0x1482) DFC1_RX_W0 */ #define MADERA_DFC1_RX_DATA_WIDTH_MASK 0x1F00 #define MADERA_DFC1_RX_DATA_WIDTH_SHIFT 8 -#define MADERA_DFC1_RX_DATA_WIDTH_WIDTH 5 #define MADERA_DFC1_RX_DATA_TYPE_MASK 0x0007 #define MADERA_DFC1_RX_DATA_TYPE_SHIFT 0 -#define MADERA_DFC1_RX_DATA_TYPE_WIDTH 3 /* (0x1484) DFC1_TX_W0 */ #define MADERA_DFC1_TX_DATA_WIDTH_MASK 0x1F00 #define MADERA_DFC1_TX_DATA_WIDTH_SHIFT 8 -#define MADERA_DFC1_TX_DATA_WIDTH_WIDTH 5 #define MADERA_DFC1_TX_DATA_TYPE_MASK 0x0007 #define MADERA_DFC1_TX_DATA_TYPE_SHIFT 0 -#define MADERA_DFC1_TX_DATA_TYPE_WIDTH 3 /* (0x1600) ADSP2_IRQ0 */ #define MADERA_DSP_IRQ2 0x0002 @@ -3636,449 +3103,347 @@ #define MADERA_GP1_LVL 0x8000 #define MADERA_GP1_LVL_MASK 0x8000 #define MADERA_GP1_LVL_SHIFT 15 -#define MADERA_GP1_LVL_WIDTH 1 #define MADERA_GP1_OP_CFG 0x4000 #define MADERA_GP1_OP_CFG_MASK 0x4000 #define MADERA_GP1_OP_CFG_SHIFT 14 -#define MADERA_GP1_OP_CFG_WIDTH 1 #define MADERA_GP1_DB 0x2000 #define MADERA_GP1_DB_MASK 0x2000 #define MADERA_GP1_DB_SHIFT 13 -#define MADERA_GP1_DB_WIDTH 1 #define MADERA_GP1_POL 0x1000 #define MADERA_GP1_POL_MASK 0x1000 #define MADERA_GP1_POL_SHIFT 12 -#define MADERA_GP1_POL_WIDTH 1 #define MADERA_GP1_IP_CFG 0x0800 #define MADERA_GP1_IP_CFG_MASK 0x0800 #define MADERA_GP1_IP_CFG_SHIFT 11 -#define MADERA_GP1_IP_CFG_WIDTH 1 #define MADERA_GP1_FN_MASK 0x03FF #define MADERA_GP1_FN_SHIFT 0 -#define MADERA_GP1_FN_WIDTH 10 /* (0x1701) GPIO1_CTRL_2 */ #define MADERA_GP1_DIR 0x8000 #define MADERA_GP1_DIR_MASK 0x8000 #define MADERA_GP1_DIR_SHIFT 15 -#define MADERA_GP1_DIR_WIDTH 1 #define MADERA_GP1_PU 0x4000 #define MADERA_GP1_PU_MASK 0x4000 #define MADERA_GP1_PU_SHIFT 14 -#define MADERA_GP1_PU_WIDTH 1 #define MADERA_GP1_PD 0x2000 #define MADERA_GP1_PD_MASK 0x2000 #define MADERA_GP1_PD_SHIFT 13 -#define MADERA_GP1_PD_WIDTH 1 #define MADERA_GP1_DRV_STR_MASK 0x1800 #define MADERA_GP1_DRV_STR_SHIFT 11 -#define MADERA_GP1_DRV_STR_WIDTH 2 /* (0x1800) IRQ1_Status_1 */ #define MADERA_CTRLIF_ERR_EINT1 0x1000 #define MADERA_CTRLIF_ERR_EINT1_MASK 0x1000 #define MADERA_CTRLIF_ERR_EINT1_SHIFT 12 -#define MADERA_CTRLIF_ERR_EINT1_WIDTH 1 #define MADERA_SYSCLK_FAIL_EINT1 0x0200 #define MADERA_SYSCLK_FAIL_EINT1_MASK 0x0200 #define MADERA_SYSCLK_FAIL_EINT1_SHIFT 9 -#define MADERA_SYSCLK_FAIL_EINT1_WIDTH 1 #define MADERA_CLOCK_DETECT_EINT1 0x0100 #define MADERA_CLOCK_DETECT_EINT1_MASK 0x0100 #define MADERA_CLOCK_DETECT_EINT1_SHIFT 8 -#define MADERA_CLOCK_DETECT_EINT1_WIDTH 1 #define MADERA_BOOT_DONE_EINT1 0x0080 #define MADERA_BOOT_DONE_EINT1_MASK 0x0080 #define MADERA_BOOT_DONE_EINT1_SHIFT 7 -#define MADERA_BOOT_DONE_EINT1_WIDTH 1 /* (0x1801) IRQ1_Status_2 */ #define MADERA_FLLAO_LOCK_EINT1 0x0800 #define MADERA_FLLAO_LOCK_EINT1_MASK 0x0800 #define MADERA_FLLAO_LOCK_EINT1_SHIFT 11 -#define MADERA_FLLAO_LOCK_EINT1_WIDTH 1 #define MADERA_FLL3_LOCK_EINT1 0x0400 #define MADERA_FLL3_LOCK_EINT1_MASK 0x0400 #define MADERA_FLL3_LOCK_EINT1_SHIFT 10 -#define MADERA_FLL3_LOCK_EINT1_WIDTH 1 #define MADERA_FLL2_LOCK_EINT1 0x0200 #define MADERA_FLL2_LOCK_EINT1_MASK 0x0200 #define MADERA_FLL2_LOCK_EINT1_SHIFT 9 -#define MADERA_FLL2_LOCK_EINT1_WIDTH 1 #define MADERA_FLL1_LOCK_EINT1 0x0100 #define MADERA_FLL1_LOCK_EINT1_MASK 0x0100 #define MADERA_FLL1_LOCK_EINT1_SHIFT 8 -#define MADERA_FLL1_LOCK_EINT1_WIDTH 1 /* (0x1805) IRQ1_Status_6 */ #define MADERA_MICDET2_EINT1 0x0200 #define MADERA_MICDET2_EINT1_MASK 0x0200 #define MADERA_MICDET2_EINT1_SHIFT 9 -#define MADERA_MICDET2_EINT1_WIDTH 1 #define MADERA_MICDET1_EINT1 0x0100 #define MADERA_MICDET1_EINT1_MASK 0x0100 #define MADERA_MICDET1_EINT1_SHIFT 8 -#define MADERA_MICDET1_EINT1_WIDTH 1 #define MADERA_HPDET_EINT1 0x0001 #define MADERA_HPDET_EINT1_MASK 0x0001 #define MADERA_HPDET_EINT1_SHIFT 0 -#define MADERA_HPDET_EINT1_WIDTH 1 /* (0x1806) IRQ1_Status_7 */ #define MADERA_MICD_CLAMP_FALL_EINT1 0x0020 #define MADERA_MICD_CLAMP_FALL_EINT1_MASK 0x0020 #define MADERA_MICD_CLAMP_FALL_EINT1_SHIFT 5 -#define MADERA_MICD_CLAMP_FALL_EINT1_WIDTH 1 #define MADERA_MICD_CLAMP_RISE_EINT1 0x0010 #define MADERA_MICD_CLAMP_RISE_EINT1_MASK 0x0010 #define MADERA_MICD_CLAMP_RISE_EINT1_SHIFT 4 -#define MADERA_MICD_CLAMP_RISE_EINT1_WIDTH 1 #define MADERA_JD2_FALL_EINT1 0x0008 #define MADERA_JD2_FALL_EINT1_MASK 0x0008 #define MADERA_JD2_FALL_EINT1_SHIFT 3 -#define MADERA_JD2_FALL_EINT1_WIDTH 1 #define MADERA_JD2_RISE_EINT1 0x0004 #define MADERA_JD2_RISE_EINT1_MASK 0x0004 #define MADERA_JD2_RISE_EINT1_SHIFT 2 -#define MADERA_JD2_RISE_EINT1_WIDTH 1 #define MADERA_JD1_FALL_EINT1 0x0002 #define MADERA_JD1_FALL_EINT1_MASK 0x0002 #define MADERA_JD1_FALL_EINT1_SHIFT 1 -#define MADERA_JD1_FALL_EINT1_WIDTH 1 #define MADERA_JD1_RISE_EINT1 0x0001 #define MADERA_JD1_RISE_EINT1_MASK 0x0001 #define MADERA_JD1_RISE_EINT1_SHIFT 0 -#define MADERA_JD1_RISE_EINT1_WIDTH 1 /* (0x1808) IRQ1_Status_9 */ #define MADERA_ASRC2_IN2_LOCK_EINT1 0x0800 #define MADERA_ASRC2_IN2_LOCK_EINT1_MASK 0x0800 #define MADERA_ASRC2_IN2_LOCK_EINT1_SHIFT 11 -#define MADERA_ASRC2_IN2_LOCK_EINT1_WIDTH 1 #define MADERA_ASRC2_IN1_LOCK_EINT1 0x0400 #define MADERA_ASRC2_IN1_LOCK_EINT1_MASK 0x0400 #define MADERA_ASRC2_IN1_LOCK_EINT1_SHIFT 10 -#define MADERA_ASRC2_IN1_LOCK_EINT1_WIDTH 1 #define MADERA_ASRC1_IN2_LOCK_EINT1 0x0200 #define MADERA_ASRC1_IN2_LOCK_EINT1_MASK 0x0200 #define MADERA_ASRC1_IN2_LOCK_EINT1_SHIFT 9 -#define MADERA_ASRC1_IN2_LOCK_EINT1_WIDTH 1 #define MADERA_ASRC1_IN1_LOCK_EINT1 0x0100 #define MADERA_ASRC1_IN1_LOCK_EINT1_MASK 0x0100 #define MADERA_ASRC1_IN1_LOCK_EINT1_SHIFT 8 -#define MADERA_ASRC1_IN1_LOCK_EINT1_WIDTH 1 #define MADERA_DRC2_SIG_DET_EINT1 0x0002 #define MADERA_DRC2_SIG_DET_EINT1_MASK 0x0002 #define MADERA_DRC2_SIG_DET_EINT1_SHIFT 1 -#define MADERA_DRC2_SIG_DET_EINT1_WIDTH 1 #define MADERA_DRC1_SIG_DET_EINT1 0x0001 #define MADERA_DRC1_SIG_DET_EINT1_MASK 0x0001 #define MADERA_DRC1_SIG_DET_EINT1_SHIFT 0 -#define MADERA_DRC1_SIG_DET_EINT1_WIDTH 1 /* (0x180A) IRQ1_Status_11 */ #define MADERA_DSP_IRQ16_EINT1 0x8000 #define MADERA_DSP_IRQ16_EINT1_MASK 0x8000 #define MADERA_DSP_IRQ16_EINT1_SHIFT 15 -#define MADERA_DSP_IRQ16_EINT1_WIDTH 1 #define MADERA_DSP_IRQ15_EINT1 0x4000 #define MADERA_DSP_IRQ15_EINT1_MASK 0x4000 #define MADERA_DSP_IRQ15_EINT1_SHIFT 14 -#define MADERA_DSP_IRQ15_EINT1_WIDTH 1 #define MADERA_DSP_IRQ14_EINT1 0x2000 #define MADERA_DSP_IRQ14_EINT1_MASK 0x2000 #define MADERA_DSP_IRQ14_EINT1_SHIFT 13 -#define MADERA_DSP_IRQ14_EINT1_WIDTH 1 #define MADERA_DSP_IRQ13_EINT1 0x1000 #define MADERA_DSP_IRQ13_EINT1_MASK 0x1000 #define MADERA_DSP_IRQ13_EINT1_SHIFT 12 -#define MADERA_DSP_IRQ13_EINT1_WIDTH 1 #define MADERA_DSP_IRQ12_EINT1 0x0800 #define MADERA_DSP_IRQ12_EINT1_MASK 0x0800 #define MADERA_DSP_IRQ12_EINT1_SHIFT 11 -#define MADERA_DSP_IRQ12_EINT1_WIDTH 1 #define MADERA_DSP_IRQ11_EINT1 0x0400 #define MADERA_DSP_IRQ11_EINT1_MASK 0x0400 #define MADERA_DSP_IRQ11_EINT1_SHIFT 10 -#define MADERA_DSP_IRQ11_EINT1_WIDTH 1 #define MADERA_DSP_IRQ10_EINT1 0x0200 #define MADERA_DSP_IRQ10_EINT1_MASK 0x0200 #define MADERA_DSP_IRQ10_EINT1_SHIFT 9 -#define MADERA_DSP_IRQ10_EINT1_WIDTH 1 #define MADERA_DSP_IRQ9_EINT1 0x0100 #define MADERA_DSP_IRQ9_EINT1_MASK 0x0100 #define MADERA_DSP_IRQ9_EINT1_SHIFT 8 -#define MADERA_DSP_IRQ9_EINT1_WIDTH 1 #define MADERA_DSP_IRQ8_EINT1 0x0080 #define MADERA_DSP_IRQ8_EINT1_MASK 0x0080 #define MADERA_DSP_IRQ8_EINT1_SHIFT 7 -#define MADERA_DSP_IRQ8_EINT1_WIDTH 1 #define MADERA_DSP_IRQ7_EINT1 0x0040 #define MADERA_DSP_IRQ7_EINT1_MASK 0x0040 #define MADERA_DSP_IRQ7_EINT1_SHIFT 6 -#define MADERA_DSP_IRQ7_EINT1_WIDTH 1 #define MADERA_DSP_IRQ6_EINT1 0x0020 #define MADERA_DSP_IRQ6_EINT1_MASK 0x0020 #define MADERA_DSP_IRQ6_EINT1_SHIFT 5 -#define MADERA_DSP_IRQ6_EINT1_WIDTH 1 #define MADERA_DSP_IRQ5_EINT1 0x0010 #define MADERA_DSP_IRQ5_EINT1_MASK 0x0010 #define MADERA_DSP_IRQ5_EINT1_SHIFT 4 -#define MADERA_DSP_IRQ5_EINT1_WIDTH 1 #define MADERA_DSP_IRQ4_EINT1 0x0008 #define MADERA_DSP_IRQ4_EINT1_MASK 0x0008 #define MADERA_DSP_IRQ4_EINT1_SHIFT 3 -#define MADERA_DSP_IRQ4_EINT1_WIDTH 1 #define MADERA_DSP_IRQ3_EINT1 0x0004 #define MADERA_DSP_IRQ3_EINT1_MASK 0x0004 #define MADERA_DSP_IRQ3_EINT1_SHIFT 2 -#define MADERA_DSP_IRQ3_EINT1_WIDTH 1 #define MADERA_DSP_IRQ2_EINT1 0x0002 #define MADERA_DSP_IRQ2_EINT1_MASK 0x0002 #define MADERA_DSP_IRQ2_EINT1_SHIFT 1 -#define MADERA_DSP_IRQ2_EINT1_WIDTH 1 #define MADERA_DSP_IRQ1_EINT1 0x0001 #define MADERA_DSP_IRQ1_EINT1_MASK 0x0001 #define MADERA_DSP_IRQ1_EINT1_SHIFT 0 -#define MADERA_DSP_IRQ1_EINT1_WIDTH 1 /* (0x180B) IRQ1_Status_12 */ #define MADERA_SPKOUTR_SC_EINT1 0x0080 #define MADERA_SPKOUTR_SC_EINT1_MASK 0x0080 #define MADERA_SPKOUTR_SC_EINT1_SHIFT 7 -#define MADERA_SPKOUTR_SC_EINT1_WIDTH 1 #define MADERA_SPKOUTL_SC_EINT1 0x0040 #define MADERA_SPKOUTL_SC_EINT1_MASK 0x0040 #define MADERA_SPKOUTL_SC_EINT1_SHIFT 6 -#define MADERA_SPKOUTL_SC_EINT1_WIDTH 1 #define MADERA_HP3R_SC_EINT1 0x0020 #define MADERA_HP3R_SC_EINT1_MASK 0x0020 #define MADERA_HP3R_SC_EINT1_SHIFT 5 -#define MADERA_HP3R_SC_EINT1_WIDTH 1 #define MADERA_HP3L_SC_EINT1 0x0010 #define MADERA_HP3L_SC_EINT1_MASK 0x0010 #define MADERA_HP3L_SC_EINT1_SHIFT 4 -#define MADERA_HP3L_SC_EINT1_WIDTH 1 #define MADERA_HP2R_SC_EINT1 0x0008 #define MADERA_HP2R_SC_EINT1_MASK 0x0008 #define MADERA_HP2R_SC_EINT1_SHIFT 3 -#define MADERA_HP2R_SC_EINT1_WIDTH 1 #define MADERA_HP2L_SC_EINT1 0x0004 #define MADERA_HP2L_SC_EINT1_MASK 0x0004 #define MADERA_HP2L_SC_EINT1_SHIFT 2 -#define MADERA_HP2L_SC_EINT1_WIDTH 1 #define MADERA_HP1R_SC_EINT1 0x0002 #define MADERA_HP1R_SC_EINT1_MASK 0x0002 #define MADERA_HP1R_SC_EINT1_SHIFT 1 -#define MADERA_HP1R_SC_EINT1_WIDTH 1 #define MADERA_HP1L_SC_EINT1 0x0001 #define MADERA_HP1L_SC_EINT1_MASK 0x0001 #define MADERA_HP1L_SC_EINT1_SHIFT 0 -#define MADERA_HP1L_SC_EINT1_WIDTH 1 /* (0x180E) IRQ1_Status_15 */ #define MADERA_SPK_OVERHEAT_WARN_EINT1 0x0004 #define MADERA_SPK_OVERHEAT_WARN_EINT1_MASK 0x0004 #define MADERA_SPK_OVERHEAT_WARN_EINT1_SHIFT 2 -#define MADERA_SPK_OVERHEAT_WARN_EINT1_WIDTH 1 #define MADERA_SPK_OVERHEAT_EINT1 0x0002 #define MADERA_SPK_OVERHEAT_EINT1_MASK 0x0002 #define MADERA_SPK_OVERHEAT_EINT1_SHIFT 1 -#define MADERA_SPK_OVERHEAT_EINT1_WIDTH 1 #define MADERA_SPK_SHUTDOWN_EINT1 0x0001 #define MADERA_SPK_SHUTDOWN_EINT1_MASK 0x0001 #define MADERA_SPK_SHUTDOWN_EINT1_SHIFT 0 -#define MADERA_SPK_SHUTDOWN_EINT1_WIDTH 1 /* (0x1820) - IRQ1 Status 33 */ #define MADERA_DSP7_BUS_ERR_EINT1 0x0040 #define MADERA_DSP7_BUS_ERR_EINT1_MASK 0x0040 #define MADERA_DSP7_BUS_ERR_EINT1_SHIFT 6 -#define MADERA_DSP7_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP6_BUS_ERR_EINT1 0x0020 #define MADERA_DSP6_BUS_ERR_EINT1_MASK 0x0020 #define MADERA_DSP6_BUS_ERR_EINT1_SHIFT 5 -#define MADERA_DSP6_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP5_BUS_ERR_EINT1 0x0010 #define MADERA_DSP5_BUS_ERR_EINT1_MASK 0x0010 #define MADERA_DSP5_BUS_ERR_EINT1_SHIFT 4 -#define MADERA_DSP5_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP4_BUS_ERR_EINT1 0x0008 #define MADERA_DSP4_BUS_ERR_EINT1_MASK 0x0008 #define MADERA_DSP4_BUS_ERR_EINT1_SHIFT 3 -#define MADERA_DSP4_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP3_BUS_ERR_EINT1 0x0004 #define MADERA_DSP3_BUS_ERR_EINT1_MASK 0x0004 #define MADERA_DSP3_BUS_ERR_EINT1_SHIFT 2 -#define MADERA_DSP3_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP2_BUS_ERR_EINT1 0x0002 #define MADERA_DSP2_BUS_ERR_EINT1_MASK 0x0002 #define MADERA_DSP2_BUS_ERR_EINT1_SHIFT 1 -#define MADERA_DSP2_BUS_ERR_EINT1_WIDTH 1 #define MADERA_DSP1_BUS_ERR_EINT1 0x0001 #define MADERA_DSP1_BUS_ERR_EINT1_MASK 0x0001 #define MADERA_DSP1_BUS_ERR_EINT1_SHIFT 0 -#define MADERA_DSP1_BUS_ERR_EINT1_WIDTH 1 /* (0x1845) IRQ1_Mask_6 */ #define MADERA_IM_MICDET2_EINT1 0x0200 #define MADERA_IM_MICDET2_EINT1_MASK 0x0200 #define MADERA_IM_MICDET2_EINT1_SHIFT 9 -#define MADERA_IM_MICDET2_EINT1_WIDTH 1 #define MADERA_IM_MICDET1_EINT1 0x0100 #define MADERA_IM_MICDET1_EINT1_MASK 0x0100 #define MADERA_IM_MICDET1_EINT1_SHIFT 8 -#define MADERA_IM_MICDET1_EINT1_WIDTH 1 #define MADERA_IM_HPDET_EINT1 0x0001 #define MADERA_IM_HPDET_EINT1_MASK 0x0001 #define MADERA_IM_HPDET_EINT1_SHIFT 0 -#define MADERA_IM_HPDET_EINT1_WIDTH 1 /* (0x184E) IRQ1_Mask_15 */ #define MADERA_IM_SPK_OVERHEAT_WARN_EINT1 0x0004 #define MADERA_IM_SPK_OVERHEAT_WARN_EINT1_MASK 0x0004 #define MADERA_IM_SPK_OVERHEAT_WARN_EINT1_SHIFT 2 -#define MADERA_IM_SPK_OVERHEAT_WARN_EINT1_WIDTH 1 #define MADERA_IM_SPK_OVERHEAT_EINT1 0x0002 #define MADERA_IM_SPK_OVERHEAT_EINT1_MASK 0x0002 #define MADERA_IM_SPK_OVERHEAT_EINT1_SHIFT 1 -#define MADERA_IM_SPK_OVERHEAT_EINT1_WIDTH 1 #define MADERA_IM_SPK_SHUTDOWN_EINT1 0x0001 #define MADERA_IM_SPK_SHUTDOWN_EINT1_MASK 0x0001 #define MADERA_IM_SPK_SHUTDOWN_EINT1_SHIFT 0 -#define MADERA_IM_SPK_SHUTDOWN_EINT1_WIDTH 1 /* (0x1880) - IRQ1 Raw Status 1 */ #define MADERA_CTRLIF_ERR_STS1 0x1000 #define MADERA_CTRLIF_ERR_STS1_MASK 0x1000 #define MADERA_CTRLIF_ERR_STS1_SHIFT 12 -#define MADERA_CTRLIF_ERR_STS1_WIDTH 1 #define MADERA_SYSCLK_FAIL_STS1 0x0200 #define MADERA_SYSCLK_FAIL_STS1_MASK 0x0200 #define MADERA_SYSCLK_FAIL_STS1_SHIFT 9 -#define MADERA_SYSCLK_FAIL_STS1_WIDTH 1 #define MADERA_CLOCK_DETECT_STS1 0x0100 #define MADERA_CLOCK_DETECT_STS1_MASK 0x0100 #define MADERA_CLOCK_DETECT_STS1_SHIFT 8 -#define MADERA_CLOCK_DETECT_STS1_WIDTH 1 #define MADERA_BOOT_DONE_STS1 0x0080 #define MADERA_BOOT_DONE_STS1_MASK 0x0080 #define MADERA_BOOT_DONE_STS1_SHIFT 7 -#define MADERA_BOOT_DONE_STS1_WIDTH 1 /* (0x1881) - IRQ1 Raw Status 2 */ #define MADERA_FLL3_LOCK_STS1 0x0400 #define MADERA_FLL3_LOCK_STS1_MASK 0x0400 #define MADERA_FLL3_LOCK_STS1_SHIFT 10 -#define MADERA_FLL3_LOCK_STS1_WIDTH 1 #define MADERA_FLL2_LOCK_STS1 0x0200 #define MADERA_FLL2_LOCK_STS1_MASK 0x0200 #define MADERA_FLL2_LOCK_STS1_SHIFT 9 -#define MADERA_FLL2_LOCK_STS1_WIDTH 1 #define MADERA_FLL1_LOCK_STS1 0x0100 #define MADERA_FLL1_LOCK_STS1_MASK 0x0100 #define MADERA_FLL1_LOCK_STS1_SHIFT 8 -#define MADERA_FLL1_LOCK_STS1_WIDTH 1 /* (0x1886) - IRQ1 Raw Status 7 */ #define MADERA_MICD_CLAMP_FALL_STS1 0x0020 #define MADERA_MICD_CLAMP_FALL_STS1_MASK 0x0020 #define MADERA_MICD_CLAMP_FALL_STS1_SHIFT 5 -#define MADERA_MICD_CLAMP_FALL_STS1_WIDTH 1 #define MADERA_MICD_CLAMP_RISE_STS1 0x0010 #define MADERA_MICD_CLAMP_RISE_STS1_MASK 0x0010 #define MADERA_MICD_CLAMP_RISE_STS1_SHIFT 4 -#define MADERA_MICD_CLAMP_RISE_STS1_WIDTH 1 #define MADERA_JD2_FALL_STS1 0x0008 #define MADERA_JD2_FALL_STS1_MASK 0x0008 #define MADERA_JD2_FALL_STS1_SHIFT 3 -#define MADERA_JD2_FALL_STS1_WIDTH 1 #define MADERA_JD2_RISE_STS1 0x0004 #define MADERA_JD2_RISE_STS1_MASK 0x0004 #define MADERA_JD2_RISE_STS1_SHIFT 2 -#define MADERA_JD2_RISE_STS1_WIDTH 1 #define MADERA_JD1_FALL_STS1 0x0002 #define MADERA_JD1_FALL_STS1_MASK 0x0002 #define MADERA_JD1_FALL_STS1_SHIFT 1 -#define MADERA_JD1_FALL_STS1_WIDTH 1 #define MADERA_JD1_RISE_STS1 0x0001 #define MADERA_JD1_RISE_STS1_MASK 0x0001 #define MADERA_JD1_RISE_STS1_SHIFT 0 -#define MADERA_JD1_RISE_STS1_WIDTH 1 /* (0x188E) - IRQ1 Raw Status 15 */ #define MADERA_SPK_OVERHEAT_WARN_STS1 0x0004 #define MADERA_SPK_OVERHEAT_WARN_STS1_MASK 0x0004 #define MADERA_SPK_OVERHEAT_WARN_STS1_SHIFT 2 -#define MADERA_SPK_OVERHEAT_WARN_STS1_WIDTH 1 #define MADERA_SPK_OVERHEAT_STS1 0x0002 #define MADERA_SPK_OVERHEAT_STS1_MASK 0x0002 #define MADERA_SPK_OVERHEAT_STS1_SHIFT 1 -#define MADERA_SPK_OVERHEAT_STS1_WIDTH 1 #define MADERA_SPK_SHUTDOWN_STS1 0x0001 #define MADERA_SPK_SHUTDOWN_STS1_MASK 0x0001 #define MADERA_SPK_SHUTDOWN_STS1_SHIFT 0 -#define MADERA_SPK_SHUTDOWN_STS1_WIDTH 1 /* (0x1A06) Interrupt_Debounce_7 */ #define MADERA_MICD_CLAMP_DB 0x0010 #define MADERA_MICD_CLAMP_DB_MASK 0x0010 #define MADERA_MICD_CLAMP_DB_SHIFT 4 -#define MADERA_MICD_CLAMP_DB_WIDTH 1 #define MADERA_JD2_DB 0x0004 #define MADERA_JD2_DB_MASK 0x0004 #define MADERA_JD2_DB_SHIFT 2 -#define MADERA_JD2_DB_WIDTH 1 #define MADERA_JD1_DB 0x0001 #define MADERA_JD1_DB_MASK 0x0001 #define MADERA_JD1_DB_SHIFT 0 -#define MADERA_JD1_DB_WIDTH 1 /* (0x1A0E) Interrupt_Debounce_15 */ #define MADERA_SPK_OVERHEAT_WARN_DB 0x0004 #define MADERA_SPK_OVERHEAT_WARN_DB_MASK 0x0004 #define MADERA_SPK_OVERHEAT_WARN_DB_SHIFT 2 -#define MADERA_SPK_OVERHEAT_WARN_DB_WIDTH 1 #define MADERA_SPK_OVERHEAT_DB 0x0002 #define MADERA_SPK_OVERHEAT_DB_MASK 0x0002 #define MADERA_SPK_OVERHEAT_DB_SHIFT 1 -#define MADERA_SPK_OVERHEAT_DB_WIDTH 1 /* (0x1A80) IRQ1_CTRL */ #define MADERA_IM_IRQ1 0x0800 #define MADERA_IM_IRQ1_MASK 0x0800 #define MADERA_IM_IRQ1_SHIFT 11 -#define MADERA_IM_IRQ1_WIDTH 1 #define MADERA_IRQ_POL 0x0400 #define MADERA_IRQ_POL_MASK 0x0400 #define MADERA_IRQ_POL_SHIFT 10 -#define MADERA_IRQ_POL_WIDTH 1 /* (0x20004) OTP_HPDET_Cal_1 */ #define MADERA_OTP_HPDET_CALIB_OFFSET_11 0xFF000000 #define MADERA_OTP_HPDET_CALIB_OFFSET_11_MASK 0xFF000000 #define MADERA_OTP_HPDET_CALIB_OFFSET_11_SHIFT 24 -#define MADERA_OTP_HPDET_CALIB_OFFSET_11_WIDTH 8 #define MADERA_OTP_HPDET_CALIB_OFFSET_10 0x00FF0000 #define MADERA_OTP_HPDET_CALIB_OFFSET_10_MASK 0x00FF0000 #define MADERA_OTP_HPDET_CALIB_OFFSET_10_SHIFT 16 -#define MADERA_OTP_HPDET_CALIB_OFFSET_10_WIDTH 8 #define MADERA_OTP_HPDET_CALIB_OFFSET_01 0x0000FF00 #define MADERA_OTP_HPDET_CALIB_OFFSET_01_MASK 0x0000FF00 #define MADERA_OTP_HPDET_CALIB_OFFSET_01_SHIFT 8 -#define MADERA_OTP_HPDET_CALIB_OFFSET_01_WIDTH 8 #define MADERA_OTP_HPDET_CALIB_OFFSET_00 0x000000FF #define MADERA_OTP_HPDET_CALIB_OFFSET_00_MASK 0x000000FF #define MADERA_OTP_HPDET_CALIB_OFFSET_00_SHIFT 0 -#define MADERA_OTP_HPDET_CALIB_OFFSET_00_WIDTH 8 /* (0x20006) OTP_HPDET_Cal_2 */ #define MADERA_OTP_HPDET_GRADIENT_1X 0x0000FF00 #define MADERA_OTP_HPDET_GRADIENT_1X_MASK 0x0000FF00 #define MADERA_OTP_HPDET_GRADIENT_1X_SHIFT 8 -#define MADERA_OTP_HPDET_GRADIENT_1X_WIDTH 8 #define MADERA_OTP_HPDET_GRADIENT_0X 0x000000FF #define MADERA_OTP_HPDET_GRADIENT_0X_MASK 0x000000FF #define MADERA_OTP_HPDET_GRADIENT_0X_SHIFT 0 -#define MADERA_OTP_HPDET_GRADIENT_0X_WIDTH 8 #endif -- 2.20.1

2 1

Re: on-board sound on ASUS TRX-40
by apt-ghetto 04 Nov '20

04 Nov '20

Hi, With the mainboard Asus Prime TRX40-Pro I do not have any on-board sound. I tried to fix it myself after reading here on the mailing list (thanks to Dan Aloni and Takashi Iwai) and with the patch below, I was able to partially fix it. The front headphone still does not work and I did not test the microphone(s) input. But at least I can hear sound when using the C/SUB, REAR and LINE OUT at the rear. As mentioned, the patch does not solve all problems, but it is a little pass towards a system with working sound. The patch below was tested by me on Gentoo with 5.4.72 and on Arch Linux with 5.9.3. I hope, it will land in the kernel sources somehow. Please feel free to modify and use it. Because the real name is required, I cannot submit the patch myself. Thank you for reading this and eventually considering to submit this to the kernel sources. ----------- Begin tested patch ----------- diff -uprN -X linux-5.4.74-vanilla/Documentation/dontdiff linux-5.4.74-vanilla/sound/usb/mixer_maps.c linux-5.4.74/sound/usb/mixer_maps.c --- linux-5.4.74-vanilla/sound/usb/mixer_maps.c 2020-11-01 12:01:07.000000000 +0100 +++ linux-5.4.74/sound/usb/mixer_maps.c 2020-11-03 11:03:15.444424941 +0100 @@ -538,6 +538,11 @@ static struct usbmix_ctl_map usbmix_ctl_ .id = USB_ID(0x0b05, 0x1917), .map = asus_rog_map, }, + { /* ASUS Prime TRX40-Pro */ + .id = USB_ID(0x0b05, 0x1918), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, diff -uprN -X linux-5.4.74-vanilla/Documentation/dontdiff linux-5.4.74-vanilla/sound/usb/quirks-table.h linux-5.4.74/sound/usb/quirks-table.h --- linux-5.4.74-vanilla/sound/usb/quirks-table.h 2020-11-03 10:53:24.000396963 +0100 +++ linux-5.4.74/sound/usb/quirks-table.h 2020-11-03 10:59:02.699412985 +0100 @@ -3667,6 +3667,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge } \ } ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ +ALC1220_VB_DESKTOP(0x0b05, 0x1918), /* ASUS Prime TRX40-Pro */ ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ ----------- End tested patch ----------- Here is the version for the master branch, but totally untested: ----------- Begin untested patch from master branch ----------- diff --git a/sound/usb/card.c b/sound/usb/card.c index fa764b61fe9c..b6c1ffb65f69 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -393,6 +393,10 @@ static const struct usb_audio_device_name usb_audio_names[] = { DEVICE_NAME(0x0d8c, 0x0102, NULL, "ICUSBAUDIO7D"), DEVICE_NAME(0x0d8c, 0x0103, NULL, "Audio Advantage MicroII"), + /* ASUS Prime TRX40-Pro */ + PROFILE_NAME(0x0b05, 0x1918, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* MSI TRX40 Creator */ PROFILE_NAME(0x0db0, 0x0d64, "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index c369c81e74c4..6323faf49d67 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -563,6 +563,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0b05, 0x1917), .map = asus_rog_map, }, + { /* ASUS Prime TRX40-Pro */ + .id = USB_ID(0x0b05, 0x1918), + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, + }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), .map = trx40_mobo_map, ----------- End untested patch from master branch ----------- Sent with ProtonMail Secure Email.

1 0

[PATCH v3 0/9] ASoC: Mediatek: Add support for MT8192 SoC
by Jiaxin Yu 04 Nov '20

04 Nov '20

This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same time, the calibration function of MT6359 is completed with real machine driver. The patch is based on broonie tree "for-next" branch. Change since v2: - split the dai driver files as a seperate patch - fix dt-bindings to GPL-2.0-only License - remove unnecessary preperty descriptions such as 'maxItems' Change since v1: - fixed some typos related to dt-bindings in [v1,3/5] and [v1,5/5] - add vendor prefix to the properties, such as "mediatek,apmixedsys" - add a dependency description to indicate the required header files Jiaxin Yu (9): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver ASoC: mediatek: mt8192: support i2s in platform driver ASoC: mediatek: mt8192: support add in platform driver ASoC: mediatek: mt8192: support pcm in platform driver ASoC: mediatek: mt8192: support tdm in platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document .../bindings/sound/mt8192-afe-pcm.yaml | 100 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 42 + sound/soc/codecs/mt6359.c | 110 + sound/soc/codecs/mt6359.h | 7 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 ++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 + .../soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 306 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1489 ++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2139 +++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 ++++ .../mediatek/mt8192/mt8192-interconnection.h | 65 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 1058 ++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 +++++++++++++++++ 23 files changed, 13338 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h -- 2.18.0

2 13

ASUS TM420 headphone enable
by Kailang 04 Nov '20

04 Nov '20

Hi Takashi, ASUS TM420 need to use COEF bit to turn on HP. Attach was enable patch. BR, Kailang

2 2

[PATCH v2 1/1] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 04 Nov '20

04 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c1fec932c49d..3f4e14cdd15a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -709,11 +709,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -721,11 +734,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -768,6 +795,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

1 0

[PATCH v2 0/1] allow pause and resume during draining
by Gyeongtaek Lee 04 Nov '20

04 Nov '20

On Mon, 26 Oct 2020 18:01:33 +0100, Takashi Iwai wrote: >On Mon, 26 Oct 2020 10:18:38 +0100, >Gyeongtaek Lee wrote: >> >> With a stream with low bitrate, user can't pause or resume the stream >> near the end of the stream because current ALSA doesn't allow it. >> If the stream has very low bitrate enough to store whole stream into >> the buffer, user can't do anything except stop the stream and then >> restart it from the first because most of applications call draining >> after sending last frame to the kernel. >> If pause, resume are allowed during draining, user experience can be >> enhanced. >> To prevent malfunction in HW drivers which don't support pause >> during draining, pause during draining will only work if HW driver >> enable this feature explicitly by calling >> snd_compr_use_pause_in_draining(). >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org > >Could you restart the new thread? It's been hanging too deeply and >hard to read through. > >Since it's the revised patch, please give the revision number (v2 or >such) and show what's different from the previous patches. > Ok. I'll send the patch again with [PATCH v2] prefix. >About the changes: > >> +/** >> + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state >> + * @substream: compress substream to set >> + * >> + * Allow pause and resume in draining state. >> + * Only HW driver supports this transition can call this API. >> + */ >> +static inline void snd_compr_use_pause_in_draining( >> + struct snd_compr_stream *substream) >> +{ >> + substream->device->use_pause_in_draining = true; >> +} > >How to set the flag is an open question. A natural way would be to >set it somehow at creating the component object, but currently there >seems no way to pass any flags. Could you explain more about what is your concerning? For me, calling snd_compr_use_pause_in_draining() in open() callback of snd_compr_ops was good enough. I've tested it and it worked well on linux 5.4. > > >thanks, > >Takashi > Gyeongtaek Lee (1): ALSA: compress: allow pause and resume during draining include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) base-commit: 3cea11cd5e3b00d91caf0b4730194039b45c5891 -- 2.21.0

1 0

(Regression) Broken pipe when streaming music from bandcamp with mpv
by GitHub issues - edited 04 Nov '20

04 Nov '20

alsa-project/alsa-lib issue #96 was edited from orbea: OS: `Slackware64-current` alsa-lib: `1.2.4` mpv: https://github.com/mpv-player/mpv/commit/172146e9f7a231b5de21921d883612d18b… yt-dlc: `2020.10.31` (https://github.com/blackjack4494/yt-dlc - bandcamp support is broken with youtube-dl currently) When streaming music from bandcamp.com with alsa it will fail with a broken pipe when automatically changing tracks at the end of a track. This will not happen if the track is skipped. ``` $ mpv https://actnoir1.bandcamp.com/album/shape-a-new-start Playing: https://actnoir1.bandcamp.com/track/shatterproof-beauty (+) Audio --aid=1 (mp3 2ch 44100Hz) AO: [alsa] 44100Hz stereo 2ch float A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s Playing: https://actnoir1.bandcamp.com/track/set-fire A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s [ao/alsa] snd_pcm_status: Broken pipe A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s (+) Audio --aid=1 (mp3 2ch 44100Hz) [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe mpv: ../audio/out/buffer.c:585: _Bool ao_play_data(struct ao *): Assertion `space >= 0' failed. The process was killed by SIGABRT: Aborted ``` mpv log: [mpv.log](https://github.com/alsa-project/alsa-lib/files/5485135/mpv.log) alsa-info log: [alsa-info.log](https://github.com/alsa-project/alsa-lib/files/5485141/alsa-… I bisected this and found the first bad commit. ``` $ git bisect good 4f90392f07e8822d1984ed990f622ad36022a4a3 is the first bad commit commit 4f90392f07e8822d1984ed990f622ad36022a4a3 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Oct 9 19:57:57 2020 +0200 pcm: fix the snd_pcm_plugin_status() avail and delay fields The avail and delay fields in the returned status structure does not reflect the actual hw_ptr/appl_ptr. This change correct this. TODO: Unfortunately, the delay might contain also information about extra hardware / buffering delay which is hidden with this change. Link: https://lore.kernel.org/alsa-devel/d9c1f37e-5c8d-f289-270e-c6cda7a56ce3@axi… Reported-by: Jonas Holmberg <jonashg(a)axis.com> Tested-by: Jonas Holmberg <jonashg(a)axis.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> src/pcm/pcm_plugin.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) ``` 4f90392f07e8822d1984ed990f622ad36022a4a3 Issue URL : https://github.com/alsa-project/alsa-lib/issues/96 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

(Regression) Broken pipe when streaming music from bandcamp with mpv
by GitHub issues - opened 04 Nov '20

04 Nov '20

alsa-project/alsa-lib issue #96 was opened from orbea: OS: `Slackware64-current` alsa-lib: `1.2.4` mpv: https://github.com/mpv-player/mpv/commit/172146e9f7a231b5de21921d883612d18b… yt-dlc: `2020.10.31` When streaming music from bandcamp.com with alsa it will fail with a broken pipe when automatically changing tracks at the end of a track. This will not happen if the track is skipped. ``` $ mpv https://actnoir1.bandcamp.com/album/shape-a-new-start Playing: https://actnoir1.bandcamp.com/track/shatterproof-beauty (+) Audio --aid=1 (mp3 2ch 44100Hz) AO: [alsa] 44100Hz stereo 2ch float A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s Playing: https://actnoir1.bandcamp.com/track/set-fire A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s [ao/alsa] snd_pcm_status: Broken pipe A: 00:05:20 / 00:05:21 (100%) Cache: 0.0s (+) Audio --aid=1 (mp3 2ch 44100Hz) [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe [ao/alsa] snd_pcm_status: Broken pipe mpv: ../audio/out/buffer.c:585: _Bool ao_play_data(struct ao *): Assertion `space >= 0' failed. The process was killed by SIGABRT: Aborted ``` mpv log: [mpv.log](https://github.com/alsa-project/alsa-lib/files/5485135/mpv.log) alsa-info log: [alsa-info.log](https://github.com/alsa-project/alsa-lib/files/5485141/alsa-… I bisected this and found the first bad commit. ``` $ git bisect good 4f90392f07e8822d1984ed990f622ad36022a4a3 is the first bad commit commit 4f90392f07e8822d1984ed990f622ad36022a4a3 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Fri Oct 9 19:57:57 2020 +0200 pcm: fix the snd_pcm_plugin_status() avail and delay fields The avail and delay fields in the returned status structure does not reflect the actual hw_ptr/appl_ptr. This change correct this. TODO: Unfortunately, the delay might contain also information about extra hardware / buffering delay which is hidden with this change. Link: https://lore.kernel.org/alsa-devel/d9c1f37e-5c8d-f289-270e-c6cda7a56ce3@axi… Reported-by: Jonas Holmberg <jonashg(a)axis.com> Tested-by: Jonas Holmberg <jonashg(a)axis.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> src/pcm/pcm_plugin.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) ``` 4f90392f07e8822d1984ed990f622ad36022a4a3 Issue URL : https://github.com/alsa-project/alsa-lib/issues/96 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 04 Nov '20

04 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 107 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 +++++++++++++++ 2 files changed, 149 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..b067bd175287 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,112 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 2: + val |= RT1015_I2S_TX_2CH; + break; + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 16: + val |= RT1015_I2S_CH_TX_LEN_16B; + break; + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + /* This is an assumption that the system sends stereo audio to the amplifier typically. + * And the stereo audio is placed in slot 0/2/4/6 as the starting slot. + * According to the slot number, we also set the mono LR to select the right slot. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1081,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

3 2

[PATCH] sound/core/seq: remove dead code
by Yu Hao 03 Nov '20

03 Nov '20

The function snd_seq_queue_client_termination() is only called from function seq_free_client1(). The seq_free_client1() calls function snd_seq_queue_client_leave(), which deletes all objects whose owner equals to client->number in global array queue_list, before the function snd_seq_queue_client_termination(), which checks whether there are any objects in global array queue_list whose owner equals to client->number, with the same argument client->number. So the checking code in function snd_seq_queue_client_termination() is dead code. Remove those dead code. Signed-off-by: Yu Hao <yuhaobehappy(a)gmail.com> --- sound/core/seq/seq_queue.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 71a6ea62c3be..91b3f3295d0b 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -545,21 +545,10 @@ void snd_seq_queue_client_termination(int client) unsigned long flags; int i; struct snd_seq_queue *q; - bool matched; for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { if ((q = queueptr(i)) == NULL) continue; - spin_lock_irqsave(&q->owner_lock, flags); - matched = (q->owner == client); - if (matched) - q->klocked = 1; - spin_unlock_irqrestore(&q->owner_lock, flags); - if (matched) { - if (q->timer->running) - snd_seq_timer_stop(q->timer); - snd_seq_timer_reset(q->timer); - } queuefree(q); } } -- 2.17.1

3 4

[PATCH] ASoC: rt715:add Mic Mute LED control support
by Perry Yuan 03 Nov '20

03 Nov '20

From: perry_yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing This patch allow to change micmute led state through this micphone led control interface like hda_generic provided. Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> Signed-off-by: Limonciello Mario <mario_limonciello(a)dell.com> --- sound/soc/codecs/rt715.c | 43 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 1 + 2 files changed, 44 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 099c8bd20006..2df2895d0092 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/leds.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -213,6 +214,45 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +static const char *rt715_micmute_led_mode[] = { + "Off", "On" +}; + +static const struct soc_enum rt715_micmute_led_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rt715_micmute_led_mode), + rt715_micmute_led_mode); + +static int rt715_mic_mute_led_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + int led_mode = rt715->micmute_led; + + ucontrol->value.integer.value[0] = led_mode; +#if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); +#endif + return 0; +} + +static int rt715_micmute_led_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + int led_mode = ucontrol->value.integer.value[0]; + + rt715->micmute_led = led_mode; +#if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) + ledtrig_audio_set(LED_AUDIO_MICMUTE, + rt715->micmute_led ? LED_ON : LED_OFF); +#endif + return 0; +} + + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -277,6 +317,9 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_LINE2_L, RT715_DIR_IN_SFT, 3, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, mic_vol_tlv), + /*Micmute Led Control*/ + SOC_ENUM_EXT("Micmute Led Mode", rt715_micmute_led_mode_enum, + rt715_mic_mute_led_mode_get, rt715_micmute_led_mode_put), }; static int rt715_mux_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index df0f24f9bc0c..32917b7846b4 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,7 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + int micmute_led; }; struct sdw_stream_data { -- 2.25.1

5 9

[PATCH] ASoC: ti: davinci-mcasp: remove always zero of davinci_mcasp_get_dt_params
by Zhang Qilong 03 Nov '20

03 Nov '20

davinci_mcasp_get_dt_params alway return zero, and its return value could be ignored by the caller. So make it 'void' type to avoid the check its return value. Fixes: 764958f2b5239 ("ASoC: ti: davinci-mcasp: Support for auxclk-fs-ratio") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/ti/davinci-mcasp.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 4b46dd827f3f..63c492862383 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2112,20 +2112,18 @@ static inline int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp) } #endif /* CONFIG_GPIOLIB */ -static int davinci_mcasp_get_dt_params(struct davinci_mcasp *mcasp) +static void davinci_mcasp_get_dt_params(struct davinci_mcasp *mcasp) { struct device_node *np = mcasp->dev->of_node; int ret; u32 val; if (!np) - return 0; + return; ret = of_property_read_u32(np, "auxclk-fs-ratio", &val); if (ret >= 0) mcasp->auxclk_fs_ratio = val; - - return 0; } static int davinci_mcasp_probe(struct platform_device *pdev) @@ -2361,9 +2359,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret) goto err; - ret = davinci_mcasp_get_dt_params(mcasp); - if (ret) - return -EINVAL; + davinci_mcasp_get_dt_params(mcasp); ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, -- 2.17.1

3 2

[PATCH] ASoC: TSCS42xx: remove unneeded semicolon
by trix＠redhat.com 03 Nov '20

03 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/tscs42xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 3265d3e8cb28..6200fab7896f 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -66,7 +66,7 @@ static bool tscs42xx_volatile(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tscs42xx_precious(struct device *dev, unsigned int reg) @@ -81,7 +81,7 @@ static bool tscs42xx_precious(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tscs42xx_regmap = { @@ -1294,7 +1294,7 @@ static int part_is_valid(struct tscs42xx *tscs42xx) return true; default: return false; - }; + } } static int set_sysclk(struct snd_soc_component *component) -- 2.18.1

2 1

[PATCH] ASoC: tegra: remove unneeded semicolon
by trix＠redhat.com 03 Nov '20

03 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/tegra/tegra186_dspk.c | 6 +++--- sound/soc/tegra/tegra210_dmic.c | 6 +++--- sound/soc/tegra/tegra210_i2s.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 0cbe31e2c7e9..7d9948fb2ca7 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -310,7 +310,7 @@ static bool tegra186_dspk_wr_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra186_dspk_rd_reg(struct device *dev, unsigned int reg) @@ -326,7 +326,7 @@ static bool tegra186_dspk_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra186_dspk_volatile_reg(struct device *dev, unsigned int reg) @@ -339,7 +339,7 @@ static bool tegra186_dspk_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra186_dspk_regmap = { diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index a661f40bc41c..ead2c99bf72e 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -322,7 +322,7 @@ static bool tegra210_dmic_wr_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra210_dmic_rd_reg(struct device *dev, unsigned int reg) @@ -338,7 +338,7 @@ static bool tegra210_dmic_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra210_dmic_volatile_reg(struct device *dev, unsigned int reg) @@ -353,7 +353,7 @@ static bool tegra210_dmic_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra210_dmic_regmap_config = { diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index a383bd5c51cd..ca31ec92e508 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -662,7 +662,7 @@ static bool tegra210_i2s_wr_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra210_i2s_rd_reg(struct device *dev, unsigned int reg) @@ -682,7 +682,7 @@ static bool tegra210_i2s_rd_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tegra210_i2s_volatile_reg(struct device *dev, unsigned int reg) @@ -701,7 +701,7 @@ static bool tegra210_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config tegra210_i2s_regmap_config = { -- 2.18.1

3 2

[PATCH] ASoC: TSCS454: remove unneeded semicolon
by trix＠redhat.com 03 Nov '20

03 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/tscs454.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index d0af16b4db2f..cd1f1a592386 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -177,7 +177,7 @@ static bool tscs454_volatile(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool tscs454_writable(struct device *dev, unsigned int reg) @@ -197,7 +197,7 @@ static bool tscs454_writable(struct device *dev, unsigned int reg) return false; default: return true; - }; + } } static bool tscs454_readable(struct device *dev, unsigned int reg) @@ -217,7 +217,7 @@ static bool tscs454_readable(struct device *dev, unsigned int reg) return false; default: return true; - }; + } } static bool tscs454_precious(struct device *dev, unsigned int reg) @@ -246,7 +246,7 @@ static bool tscs454_precious(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_range_cfg tscs454_regmap_range_cfg = { -- 2.18.1

2 1

[PATCH] ASoC: qcom: sc7180: Fix some indenting in sc7180_lpass_alloc_dma_channel()
by Dan Carpenter 03 Nov '20

03 Nov '20

This code is correct, but it should be indented one more tab. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/qcom/lpass-sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index c6292f9e613f..61a208fe6875 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -96,8 +96,8 @@ static int sc7180_lpass_alloc_dma_channel(struct lpass_data *drvdata, chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, v->rdma_channels); - if (chan >= v->rdma_channels) - return -EBUSY; + if (chan >= v->rdma_channels) + return -EBUSY; } else { chan = find_next_zero_bit(&drvdata->dma_ch_bit_map, v->wrdma_channel_start + -- 2.28.0

2 1

[PATCH] ASoC: atmel-i2s: do not warn if muxclk is missing
by Codrin Ciubotariu 03 Nov '20

03 Nov '20

Besides the fact that muxclk is optional, muxclk can be set using assigned-clocks, removing the need to set it in driver. The warning is thus unneeded, so we can transform it in a debug print, eventually to just reflect that muxclk was not set by the driver. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index bbe2b638abb5..232300dda548 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -563,8 +563,8 @@ static int atmel_i2s_sama5d2_mck_init(struct atmel_i2s_dev *dev, err = PTR_ERR(muxclk); if (err == -EPROBE_DEFER) return -EPROBE_DEFER; - dev_warn(dev->dev, - "failed to get the I2S clock control: %d\n", err); + dev_dbg(dev->dev, + "failed to get the I2S clock control: %d\n", err); return 0; } -- 2.25.1

2 1

snd_bt87x on Kworld KW-TV878RF
by Tamás Hámor 03 Nov '20

03 Nov '20

Hi, After reinstalling Linux (Xubuntu 20.04.1) on my old computer the sound stopped working on my Kworld KW-TV878RF tuner card. I load the bttv module with options "card=78 radio=1" and snd-bt87x with "load_all". The corresponding dmesg is below: [ 15.626992] bttv: driver version 0.9.19 loaded [ 15.626994] bttv: using 8 buffers with 2080k (520 pages) each for capture [ 15.627024] bttv: Bt8xx card found (0) [ 15.627166] bttv: 0: Bt878 (rev 17) at 0000:04:01.0, irq: 18, latency: 32, mmio: 0xf0001000 [ 15.627177] bttv: 0: using: Jetway TV/Capture JW-TV878-FBK, Kworld KW-TV878RF [card=78,insmod option] [ 15.627359] bttv: 0: tuner type=5 [ 15.908280] bttv: 0: audio absent, no audio device found! [ 16.454464] bttv: 0: Setting PLL: 28636363 => 35468950 (needs up to 100ms) [ 16.496037] bttv: PLL set ok [ 16.497200] bttv: 0: registered device video0 [ 16.497273] bttv: 0: registered device vbi0 [ 16.497357] bttv: 0: registered device radio ... [ 682.784918] snd_bt87x 0000:04:01.1: unknown card 0x878-0x00:0x00 [ 682.784919] snd_bt87x 0000:04:01.1: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 682.785084] snd_bt87x 0000:04:01.1: bt87x0: Using board 0, analog, digital (rate 32000 Hz) The lspci output is: 04:01.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) Flags: bus master, medium devsel, latency 32, IRQ 18 Memory at f0001000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data Capabilities: [4c] Power Management version 2 Kernel driver in use: bttv Kernel modules: bttv 04:01.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) Flags: bus master, medium devsel, latency 32, IRQ 18 Memory at f0000000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data Capabilities: [4c] Power Management version 2 Kernel driver in use: snd_bt87x uname -a Linux ubuntu 5.4.0-52-generic #57-Ubuntu SMP Thu Oct 15 10:57:00 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux Regards, Tamas

2 1

[PATCH] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 03 Nov '20

03 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 ++++++++++++++ sound/soc/codecs/rt1015.c | 20 +++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 4 files changed, 43 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt index fcfd02d8d32f..2c92c8cf3c7f 100644 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -8,10 +8,16 @@ Required properties: - reg : The I2C address of the device. +Optional properties: + +- realtek,realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. Example: rt1015: codec@28 { compatible = "realtek,rt1015"; reg = <0x28>; + realtek,realtek,power-up-delay-ms = <50>; }; diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..70a7538d4c89 --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay_ms; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..967193518349 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay_ms = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay_ms); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay-ms", + &rt1015->pdata.power_up_delay_ms); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

2 1

[PATCH 0/2] ASoC: stm32: i2s: add master clock provider
by Olivier Moysan 03 Nov '20

03 Nov '20

Add master clock generation support in STM32 I2S driver. Olivier Moysan (2): ASoC: dt-bindings: add mclk provider support to stm32 i2s ASoC: stm32: i2s: add master clock provider .../bindings/sound/st,stm32-i2s.yaml | 4 + sound/soc/stm/stm32_i2s.c | 310 +++++++++++++++--- 2 files changed, 270 insertions(+), 44 deletions(-) -- 2.17.1

3 5

[PATCH] ALSA: hda: prevent undefined shift in snd_hdac_ext_bus_get_link()
by Dan Carpenter 03 Nov '20

03 Nov '20

This is harmless, but the "addr" comes from the user and it could lead to a negative shift or to shift wrapping if it's too high. Fixes: 0b00a5615dc4 ("ALSA: hdac_ext: add hdac extended controller") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/hda/ext/hdac_ext_controller.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 4d060d5b1db6..b0c0ef824d7d 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -148,6 +148,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, return NULL; if (bus->idx != bus_idx) return NULL; + if (addr < 0 || addr > 31) + return NULL; list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { -- 2.28.0

2 1

[PATCH] ASoC: wcd9335: remove unneeded semicolon
by trix＠redhat.com 03 Nov '20

03 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/wcd-clsh-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c index 817d8259758c..73258e292e7e 100644 --- a/sound/soc/codecs/wcd-clsh-v2.c +++ b/sound/soc/codecs/wcd-clsh-v2.c @@ -507,7 +507,7 @@ static bool wcd_clsh_is_state_valid(int state) return true; default: return false; - }; + } } /* -- 2.18.1

2 1

[PATCH v2 0/8] slab: provide and use krealloc_array()
by Bartosz Golaszewski 03 Nov '20

03 Nov '20

From: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy brought to my attention the fact that users allocating an array of equally sized elements should check if the size multiplication doesn't overflow. This is why we have helpers like kmalloc_array(). However we don't have krealloc_array() equivalent and there are many users who do their own multiplication when calling krealloc() for arrays. This series provides krealloc_array() and uses it in a couple places. A separate series will follow adding devm_krealloc_array() which is needed in the xilinx adc driver. v1 -> v2: - added a kernel doc for krealloc_array() - mentioned krealloc et al in the docs - collected review tags Bartosz Golaszewski (8): mm: slab: provide krealloc_array() ALSA: pcm: use krealloc_array() vhost: vringh: use krealloc_array() pinctrl: use krealloc_array() edac: ghes: use krealloc_array() drm: atomic: use krealloc_array() hwtracing: intel: use krealloc_array() dma-buf: use krealloc_array() Documentation/core-api/memory-allocation.rst | 4 ++++ drivers/dma-buf/sync_file.c | 4 ++-- drivers/edac/ghes_edac.c | 4 ++-- drivers/gpu/drm/drm_atomic.c | 3 ++- drivers/hwtracing/intel_th/msu.c | 2 +- drivers/pinctrl/pinctrl-utils.c | 2 +- drivers/vhost/vringh.c | 3 ++- include/linux/slab.h | 18 ++++++++++++++++++ sound/core/pcm_lib.c | 4 ++-- 9 files changed, 34 insertions(+), 10 deletions(-) -- 2.29.1

6 14

[PATCH] ALSA: usb-audio: add usb vendor id as DSD-capable for Khadas devices
by Artem Lapkin 03 Nov '20

03 Nov '20

Khadas audio devices ( USB_ID_VENDOR 0x3353 ) have DSD-capable implementations from XMOS need add new usb vendor id for recognition Signed-off-by: Artem Lapkin <art(a)khadas.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b4fa80ef730..c989ad8052a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1800,6 +1800,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ case 0x2ab6: /* T+A devices */ + case 0x3353: /* Khadas devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) -- 2.25.1

1 0

ucm2:sof-soundwire: Add Support For Dell Privacy Mic Mute LED Control
by GitHub pull_request - opened 03 Nov '20

03 Nov '20

alsa-project/alsa-ucm-conf pull request #60 was opened from GoPerry: Dell is going to add internal microphone mute led control for Dell privacy system, which has hardware level protection for audio privacy, when mic mute state is changed, the ucm2 will help to switch the Mic Mute LED state. Signed-off-by: perry_yuan <perry_yuan(a)dell.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/60 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/60.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2:sof-soundwire: Add Support For Dell Privacy Mic Mute LED Control
by GitHub issues - opened 03 Nov '20

03 Nov '20

alsa-project/alsa-ucm-conf issue #59 was opened from GoPerry: Dell is going to add internal microphone mute led control for Dell privacy system, which has hardware level protection for audio privacy, when mic mute state is changed, the ucm2 will help to switch the Mic Mute LED state. This solution has been discussed by this FR. https://github.com/thesofproject/linux/issues/2496 This system EC which support audio privacy needs get ACK event when the LED state need to change. I add one new mixer for RT715 (local mics) to switch the mic mute led state So the UCM2 will help to call the mixer to notify the LED state is changed. The led trigger driver will emit the notification to EC through the LED driver. Then LED state controlled by the EC will be changed accordingly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/59 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 03 Nov '20

03 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> Changes from v1 to v2: - Ducumentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Changes from v2 to v3: - Machine driver: Addressed suggestions from Tzung-Bi. - move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data. - Remove redundant judgments in dmic_set(). Changes from v3 to v4: - Machine driver: Addressed suggestions from Tzung-Bi. - Remove redundant judgments in dmic_set() and dmic_get(). - Remove 1 level indent of judgment of IS_ERR(data->dmic_sel). Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 58 ++++++++++++++++++ sound/soc/qcom/sc7180.c | 61 +++++++++++++++++++ 2 files changed, 119 insertions(+) -- 2.25.1

3 4

[PATCH] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 03 Nov '20

03 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 ++++++++++++++++ 2 files changed, 143 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..5d3e269d17f5 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,106 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + case 2: + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + case 16: + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum > 1 || !rx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_PAD_DRV2, RT1015_MONO_LR_SEL_MASK, + RT1015_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1075,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

3 2

sound: PCM/I2S Digital Mic & SPDIF
by Muni Sekhar 02 Nov '20

02 Nov '20

Hello all, I’m having a Linux X86 machine and interfacing with Xilinx FPGA endpoint over PCIe. FPGA endpoint supports various audio engines like PCM/I2S ,Digital Mic & SPDIF. I’m looking for any existing linux device driver to port it for my hardware. Does the Linux kernel tree have any reference examples for these audio engines which I can use for porting it to my hardware? If so can someone please share your inputs on this. -- Thanks, Sekhar

2 1

[PATCH] ASoC: bcm2835-i2s: remove unneeded semicolon
by trix＠redhat.com 02 Nov '20

02 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/bcm/bcm2835-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index dc34fe1559c6..c2f7631e8705 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -797,7 +797,7 @@ static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) @@ -807,7 +807,7 @@ static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config bcm2835_regmap_config = { -- 2.18.1

2 1

[PATCH] ASoC: ti: davinci-evm: Remove redundant null check before clk_disable_unprepare
by Xu Wang 02 Nov '20

02 Nov '20

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/ti/davinci-evm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 105e56ab9cdc..b043a0070d20 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -46,8 +46,7 @@ static void evm_shutdown(struct snd_pcm_substream *substream) struct snd_soc_card_drvdata_davinci *drvdata = snd_soc_card_get_drvdata(soc_card); - if (drvdata->mclk) - clk_disable_unprepare(drvdata->mclk); + clk_disable_unprepare(drvdata->mclk); } static int evm_hw_params(struct snd_pcm_substream *substream, -- 2.17.1

3 2

[PATCH] ASoC: Intel: remove unneeded semicolon
by trix＠redhat.com 02 Nov '20

02 Nov '20

From: Tom Rix <trix(a)redhat.com> A semicolon is not needed after a switch statement. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/catpt/loader.c | 2 +- sound/soc/intel/catpt/pcm.c | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index e90590559185..e21e11dac000 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -186,7 +186,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) "SST Driver capabilities missing for dev_id: %x", sst->dev_id); return -EINVAL; - }; + } } void sst_process_pending_msg(struct work_struct *work) diff --git a/sound/soc/intel/catpt/loader.c b/sound/soc/intel/catpt/loader.c index 8a5f20abcadb..40c22e4bb263 100644 --- a/sound/soc/intel/catpt/loader.c +++ b/sound/soc/intel/catpt/loader.c @@ -304,7 +304,7 @@ static int catpt_load_block(struct catpt_dev *cdev, default: sram = &cdev->dram; break; - }; + } dst_addr = sram->start + blk->ram_offset; if (alloc) { diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index ba653ebea7d1..1775540ac92b 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -92,7 +92,7 @@ catpt_get_stream_template(struct snd_pcm_substream *substream) break; default: break; - }; + } return catpt_topology[type]; } @@ -419,7 +419,7 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, break; default: return 0; - }; + } list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && -- 2.18.1

3 2

[PATCH] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 02 Nov '20

02 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- include/sound/rt1015.h | 15 +++++++++++++++ sound/soc/codecs/rt1015.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 3 files changed, 37 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..bc83ac22a6bf --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..4f9d453f1603 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay", + &rt1015->pdata.power_up_delay); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

3 2

[bug report] ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver
by Dan Carpenter 02 Nov '20

02 Nov '20

Hello Viorel Suman, The patch 28564486866f: "ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver" from Oct 13, 2020, leads to the following static checker warning: sound/soc/fsl/fsl_xcvr.c:393 fsl_xcvr_en_aud_pll() warn: 'xcvr->phy_clk' not released on lines: 373. sound/soc/fsl/fsl_xcvr.c 351 static int fsl_xcvr_en_aud_pll(struct fsl_xcvr *xcvr, u32 freq) 352 { 353 struct device *dev = &xcvr->pdev->dev; 354 int ret; 355 356 clk_disable_unprepare(xcvr->phy_clk); 357 ret = clk_set_rate(xcvr->phy_clk, freq); 358 if (ret < 0) { 359 dev_err(dev, "Error while setting AUD PLL rate: %d\n", ret); 360 return ret; 361 } 362 ret = clk_prepare_enable(xcvr->phy_clk); 363 if (ret) { 364 dev_err(dev, "failed to start PHY clock: %d\n", ret); 365 return ret; 366 } 367 368 /* Release AI interface from reset */ 369 ret = regmap_write(xcvr->regmap, FSL_XCVR_PHY_AI_CTRL_SET, 370 FSL_XCVR_PHY_AI_CTRL_AI_RESETN); 371 if (ret < 0) { 372 dev_err(dev, "Error while setting IER0: %d\n", ret); clck_disable_unprepare(xcvr->phy_clk)? 373 return ret; 374 } 375 376 if (xcvr->mode == FSL_XCVR_MODE_EARC) { /* eARC mode */ 377 /* PHY: CTRL_SET: TX_DIFF_OE, PHY_EN */ 378 fsl_xcvr_ai_write(xcvr, FSL_XCVR_PHY_CTRL_SET, 379 FSL_XCVR_PHY_CTRL_TSDIFF_OE | 380 FSL_XCVR_PHY_CTRL_PHY_EN, 1); 381 /* PHY: CTRL2_SET: EARC_TX_MODE */ 382 fsl_xcvr_ai_write(xcvr, FSL_XCVR_PHY_CTRL2_SET, 383 FSL_XCVR_PHY_CTRL2_EARC_TXMS, 1); 384 } else { /* SPDIF mode */ 385 /* PHY: CTRL_SET: TX_CLK_AUD_SS | SPDIF_EN */ 386 fsl_xcvr_ai_write(xcvr, FSL_XCVR_PHY_CTRL_SET, 387 FSL_XCVR_PHY_CTRL_TX_CLK_AUD_SS | 388 FSL_XCVR_PHY_CTRL_SPDIF_EN, 1); 389 } 390 391 dev_dbg(dev, "PLL Fexp: %u\n", freq); 392 393 return 0; 394 } regards, dan carpenter

2 2

[PATCH v3 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 02 Nov '20

02 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> Changes from v1 to v2: - Ducumentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Changes from v2 to v3: - Machine driver: Addressed suggestions from Tzung-Bi. - move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data. - Remove redundant judgments in dmic_set(). Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 58 +++++++++++++++++ sound/soc/qcom/sc7180.c | 65 +++++++++++++++++++ 2 files changed, 123 insertions(+) -- 2.25.1

3 4

[PATCH v10 00/15] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 02 Nov '20

02 Nov '20

Hi, This series add H6 I2S support and the I2S node missing to support HDMI audio in different Allwinner SoC. As we first use some TDM property to make the I2S working with the simple soundcard. We have now drop this simple sound card and a proper dedicated soundcard will be introduce later. This make the title of this series wrong now but to be able to follow the previous release I keep the same name. Regards, Clement Change since v9: - fix lrck_period computation for I2S justified mode Change since v8: - move the comment near the function prototype - collect Maxime Ripard tags Change since v7: - rebase on next-20201026 - comment about slots and slot_width Change since v6: - move set_channel_cfg() in first position - convert return value to decimal Change since v5: - Drop HDMI simple soundcard - Collect Chen-Yu Tsai tags - Configure channels from 9 to 15. - Remove DMA RX for H3/H5 - Fix Documentation for H3/H5 Change since v4: - add more comment on get_wss() and set_channel_cfg() patch - merge soundcard and DAI HDMI patches Change since v3: - add Samuel Holland patch to reconfigure FIFO_TX_REG when suspend is enabled - readd inversion to H6 LRCK sun50i_h6_i2s_set_soc_fmt() - Fix get_wss() for sun4i - Add a commit to fix checkpatch warning Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (7): ASoC: sun4i-i2s: Fix lrck_period computation for I2S justified mode ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition arm64: defconfig: Enable Allwinner i2s driver dt-bindings: sound: sun4i-i2s: Document H3 with missing RX channel possibility Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add I2S1 node Marcus Cooper (4): ASoC: sun4i-i2s: Set sign extend sample ASoC: sun4i-i2s: Add 20 and 24 bit support arm64: dts: allwinner: a64: Add I2S2 node arm: dts: sunxi: h3/h5: Add I2S2 node Samuel Holland (1): ASoC: sun4i-i2s: Fix setting of FIFO modes .../sound/allwinner,sun4i-a10-i2s.yaml | 6 +- arch/arm/boot/dts/sunxi-h3-h5.dtsi | 13 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 14 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 13 + arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 388 +++++++++++++++--- 6 files changed, 378 insertions(+), 57 deletions(-) -- 2.25.1

3 22

[PATCH] pcm: set the snd_pcm_ioplug_status() tstamp field
by Jonas Holmberg 02 Nov '20

02 Nov '20

Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_ioplug.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca32..9b1b8ac3 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -115,6 +115,7 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_ioplug_hw_ptr_update(pcm); status->state = io->data->state; status->trigger_tstamp = io->trigger_tstamp; + gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_avail(pcm); status->avail_max = io->avail_max; return 0; -- 2.26.2

1 0

[PATCH v2 0/7] Add support for Rpi4b + Cirrus Lochnagar2 and CS47L15
by Richard Fitzgerald 02 Nov '20

02 Nov '20

This set of patches provides support for using the Raspberry Pi 4b with a Cirrus Logic Lochnagar 2 audio development platform plus Cirrus Logic CS47L15 codec. Patches are needed to audio-graph-card to enable support for setting the component sysclks and plls. The codec sysclks and plls cannot be placed under the clock framwork because they are I2C/SPI-connected peripherals and access to the registers would cause a nested get of the I2C/SPI bus clock. The clock framework does not support this and it would result in a deadlock. Richard Fitzgerald (7): of: base: Add of_count_phandle_with_fixed_args() ASoC: audio-graph-card: Add plls and sysclks DT bindings ASoC: audio-graph-card: Support setting component plls and sysclks ASoC: arizona: Allow codecs to be selected from kernel config ASoC: madera: Allow codecs to be selected from kernel config ARM: dts: Add dts for RPi4b + Cirrus Logic Lochnagar2 + CS47L15 MAINTAINERS: Add dts for Cirrus Logic Lochnagar on RPi4 .../bindings/sound/audio-graph-card.txt | 44 ++++ MAINTAINERS | 1 + arch/arm/boot/dts/Makefile | 1 + ...bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts | 166 +++++++++++++++ .../dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi | 201 ++++++++++++++++++ drivers/of/base.c | 73 +++++-- include/linux/of.h | 9 + include/sound/simple_card_utils.h | 24 +++ sound/soc/codecs/Kconfig | 18 +- sound/soc/generic/audio-graph-card.c | 16 +- sound/soc/generic/simple-card-utils.c | 184 ++++++++++++++++ 11 files changed, 703 insertions(+), 34 deletions(-) create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi -- 2.20.1

3 11

[PATCH v4 00/22] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 02 Nov '20

02 Nov '20

Hi, New test done by Maxime using TDM show that's LRCK is indeed inverted so I drop the patch reverted in v2. And HDMI requires an inverted LRCK so let's readd the frame-inversion in the device-tree. I have also added a patch to change set_chan_cfg. Please note that I can't test TDM and only have a Allwinner H6. So test and comment on other Allwinner chips are welcome! Regards, Clement Change since v3: - add Samuel Holland patch to reconfigure FIFO_TX_REG when suspend is enabled - readd inversion to H6 LRCK sun50i_h6_i2s_set_soc_fmt() - Fix get_wss() for sun4i - Add a commit to fix checkpatch warning Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (6): ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 arm64: defconfig: Enable Allwinner i2s driver ASoC: sun4i-i2s: fix coding-style for callback definition Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (9): ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 Ondrej Jirman (3): arm64: dts: allwinner: Enable HDMI audio on Orange Pi PC 2 ARM: dts: sun8i-h3: Enable HDMI audio on Orange Pi PC/One arm64: dts: sun50i-h6-orangepi-3: Enable HDMI audio Samuel Holland (1): ASoC: sun4i-i2s: Fix setting of FIFO modes .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-one.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 33 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 34 ++ .../dts/allwinner/sun50i-h5-orangepi-pc2.dts | 8 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + .../dts/allwinner/sun50i-h6-orangepi-3.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 33 ++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 374 +++++++++++++++--- 14 files changed, 487 insertions(+), 54 deletions(-) -- 2.25.1

4 38

No more sound through HDMI / DisplayPort with 1.2.4
by GitHub issues - opened 02 Nov '20

02 Nov '20

alsa-project/alsa-lib issue #95 was opened from fkroener: The recent update might have broken HDMI / DisyplayPort sound on my Thinkpad T14 (AMD), using ArchLinux with 5.9.2 kernel. [old-alsa.log](https://github.com/alsa-project/alsa-lib/files/5474401/old-al… has the log of pulseaudio -vvvvv after the downgrade; [new-alsa.log](https://github.com/alsa-project/alsa-lib/files/5474402/new-al… is 1.2.4 - where the device name as well changed. Strangely only after a reboot the previous alsa-lib appears to come in effect and working, see: [rebooted-alsa.log](https://github.com/alsa-project/alsa-lib/files/5474440/r… Maybe this is a bug on my side, any pointers would be highly appreciated. I'll happily supply more logs or information if necessary. Thanks. Issue URL : https://github.com/alsa-project/alsa-lib/issues/95 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: ti: davinci-mcasp: fix error handling in davinci_mcasp_probe
by Zhang Qilong 02 Nov '20

02 Nov '20

Forgetting to call pm_runtime_disable if davinci_mcasp_get_dt_params failed in function davinci_mcasp_probe, and we should go to error handling branch. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- Changelog: V2 - changed the subject and fixed the description. --- sound/soc/ti/davinci-mcasp.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 4b46dd827f3f..ec90c23c9e20 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2362,8 +2362,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; ret = davinci_mcasp_get_dt_params(mcasp); - if (ret) - return -EINVAL; + if (ret) { + ret = -EINVAL; + goto err; + } ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, -- 2.17.1

3 4

[PATCH] ASoC: ti: davinci-mcasp: fix possible reference leak in __davinci_mcasp_set_clkdiv
by Zhang Qilong 02 Nov '20

02 Nov '20

pm_runtime_get_sync() will increment pm usage counter whatever it does. Forgetting to call pm_runtime_put will result in reference leak in __davinci_mcasp_set_clkdiv, so we should fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/ti/davinci-mcasp.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a6b72ad53b43..4b46dd827f3f 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -607,6 +607,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, int div, bool explicit) { + int ret = 0; + pm_runtime_get_sync(mcasp->dev); switch (div_id) { case MCASP_CLKDIV_AUXCLK: /* MCLK divider */ @@ -644,11 +646,11 @@ static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, break; default: - return -EINVAL; + ret = -EINVAL; } pm_runtime_put(mcasp->dev); - return 0; + return ret; } static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, -- 2.17.1

2 1

[PATCH v2 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 02 Nov '20

02 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> Changes from v1 to v2: - Ducumentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 58 ++++++++++++++++++ sound/soc/qcom/sc7180.c | 59 +++++++++++++++++++ 2 files changed, 117 insertions(+) -- 2.25.1

3 6

[PATCH] ASoC: nau8315: add codec driver
by David Lin 01 Nov '20

01 Nov '20

Add codec driver for Nuvoton NAU8315. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8315.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8315.c | 166 ++++++++++++++++++ 4 files changed, 191 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt create mode 100644 sound/soc/codecs/nau8315.c diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000000..6eaec46f384c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5791b7056af6..2b6eceb3573c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MT6358 imply SND_SOC_MT6359 imply SND_SOC_MT6660 + imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 imply SND_SOC_NAU8822 @@ -1760,6 +1761,10 @@ config SND_SOC_MT6660 Select N if you don't have MT6660 on board. Select M to build this as module. +config SND_SOC_NAU8315 + tristate "Nuvoton Technology Corporation NAU8315 CODEC" + depends on GPIOLIB + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 11ce98c25d6c..144d9256d4d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,6 +129,7 @@ snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o snd-soc-mt6359-objs := mt6359.o snd-soc-mt6660-objs := mt6660.o +snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -438,6 +439,7 @@ obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o +obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(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 diff --git a/sound/soc/codecs/nau8315.c b/sound/soc/codecs/nau8315.c new file mode 100644 index 000000000000..e6bc5c0a5036 --- /dev/null +++ b/sound/soc/codecs/nau8315.c @@ -0,0 +1,166 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8315.c -- NAU8315 ALSA SoC Audio Amplifier Driver +// +// Copyright 2020 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0(a)nuvoton.com> +// +// Based on MAX98357A.c + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> + +struct nau8315_priv { + struct gpio_desc *enable; + int enpin_switch; +}; + +static int nau8315_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (!nau8315->enable) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (nau8315->enpin_switch) { + gpiod_set_value(nau8315->enable, 1); + dev_dbg(component->dev, "set enable to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(nau8315->enable, 0); + dev_dbg(component->dev, "set enable to 0"); + break; + } + + return 0; +} + +static int nau8315_enpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + nau8315->enpin_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + nau8315->enpin_switch = 0; + + return 0; +} + +static const struct snd_soc_dapm_widget nau8315_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("EN_Pin", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8315_enpin_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route nau8315_dapm_routes[] = { + {"EN_Pin", NULL, "HiFi Playback"}, + {"Speaker", NULL, "EN_Pin"}, +}; + +static const struct snd_soc_component_driver nau8315_component_driver = { + .dapm_widgets = nau8315_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8315_dapm_widgets), + .dapm_routes = nau8315_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8315_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops nau8315_dai_ops = { + .trigger = nau8315_daiops_trigger, +}; + +#define NAU8315_RATES SNDRV_PCM_RATE_8000_96000 +#define NAU8315_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver nau8315_dai_driver = { + .name = "nau8315-hifi", + .playback = { + .stream_name = "HiFi Playback", + .formats = NAU8315_FORMATS, + .rates = NAU8315_RATES, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &nau8315_dai_ops, +}; + +static int nau8315_platform_probe(struct platform_device *pdev) +{ + struct nau8315_priv *nau8315; + + nau8315 = devm_kzalloc(&pdev->dev, sizeof(*nau8315), GFP_KERNEL); + if (!nau8315) + return -ENOMEM; + + nau8315->enable = devm_gpiod_get_optional(&pdev->dev, + "enable", GPIOD_OUT_LOW); + if (IS_ERR(nau8315->enable)) + return PTR_ERR(nau8315->enable); + + dev_set_drvdata(&pdev->dev, nau8315); + + return devm_snd_soc_register_component(&pdev->dev, + &nau8315_component_driver, + &nau8315_dai_driver, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id nau8315_device_id[] = { + { .compatible = "nuvoton,nau8315" }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8315_device_id); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8315_acpi_match[] = { + { "NVTN2010", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8315_acpi_match); +#endif + +static struct platform_driver nau8315_platform_driver = { + .driver = { + .name = "nau8315", + .of_match_table = of_match_ptr(nau8315_device_id), + .acpi_match_table = ACPI_PTR(nau8315_acpi_match), + }, + .probe = nau8315_platform_probe, +}; +module_platform_driver(nau8315_platform_driver); + +MODULE_DESCRIPTION("ASoC NAU8315 Mono Class-D Amplifier Driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1 =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

3 2

AIM SC808
by Marcin Kocur 31 Oct '20

31 Oct '20

Hello all, I own this card since recently: 03:00.0 Audio device: C-Media Electronics Inc CM8888 [Oxygen Express] Subsystem: C-Media Electronics Inc HDA Controller Flags: bus master, fast devsel, latency 0, IRQ 31, IOMMU group 7 Memory at fcd00000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel and I found out it doesn't work great. Only headphone output is functioning, RCA line out does not (haven't checked the rest). I found models list in /usr/lib/modules/5.9.2-arch1-1/build/Documentation/sound/hd-audio/models.rst and tested them all using a rmmod / modprobe bash script but none of them seems to enable RCA output. Windows drivers have possibility to switch between headphone and RCA output from manufacturer's app (there is a click sound coming from electronics of the card when I do this). I never tampered with low level hardware and I'm not familiar with ALSA development but what I was thinking is that maybe an ALSA snd_hda_intel developer would be interested in receiving this card for tests? -- Pozdrawiam / Greetings Marcin Kocur █

1 0

[PATCH 0/2] ASoC: mt6359: skip first time data at the beginning of DMIC recording
by Jiaxin Yu 31 Oct '20

31 Oct '20

This patch fixed the pop noise at the beginning of DMIC recording. Jiaxin Yu (2): ASoC: mt6359: skip first time data at the beginning of DMIC recording dt-bindings: mediatek: mt8192: add ul-delay-ms property .../devicetree/bindings/sound/mt6359.yaml | 6 +++ sound/soc/codecs/mt6359.c | 38 +++++++++++++++++-- sound/soc/codecs/mt6359.h | 1 + 3 files changed, 41 insertions(+), 4 deletions(-) -- 2.18.0

4 8

Re: [linux-sunxi] Re: [PATCH v10 00/15] Add Allwinner H3/H5/H6/A64 HDMI audio
by Jernej Škrabec 30 Oct '20

30 Oct '20

Dne petek, 30. oktober 2020 ob 21:50:43 CET je Stefan Monnier napisal(a): > >> This series add H6 I2S support and the I2S node missing to support > >> HDMI audio in different Allwinner SoC. > > > > Applied to > > > > https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git > > for-next > > Yay! Note that this doesn't bring HDMI audio card just yet. Another driver will be needed for that. > > Now, I wonder: will that make it easier to add support for HDMI-Audio for > the A10/A20? No, A10/A20 HDMI audio uses completely different interface. > > (there was a patch for that submitted earlier this year by Stefan > Mavrodiev <stefan(a)olimex.com>, but it seems there hasn't been any > progress on it since then (I think the last message on it concluded that > it should be rewritten to use ALSA instead of ASoC)). IIUC original author left Olimex, so work stalled. Best regards, Jernej > > [ To clarify, don't know what's the difference between ALSA and ASoC; > I'm only interested here as an owner of an A20 box on which I'd > love to be able to use the HDMI-Audio. ] > > > -- Stefan

1 0

[PATCH v2 0/2] Add rt1015 support to CML boards
by Brent Lu 30 Oct '20

30 Oct '20

First patch adds tdm 4-slot 100fs DAI setting to avoid jitter of using 64fs on CML boards. Second patch is a DMI quirk for HP Dooly. Changes since v1: -Add comment on Dooly's DMI quirk Brent Lu (2): ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682 ASoC: intel: sof_rt5682: Add quirk for Dooly sound/soc/intel/boards/sof_rt5682.c | 65 +++++++++++++++++-- .../intel/common/soc-acpi-intel-cml-match.c | 13 ++++ 2 files changed, 73 insertions(+), 5 deletions(-) -- 2.17.1

3 4

[PATCH] ASoC: qcom: MSM8996 depends on COMMON_CLK
by Randy Dunlap 30 Oct '20

30 Oct '20

From: Randy Dunlap <rdunlap(a)infradead.org> Fix a kconfig warning in sound/soc/qcom: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=y] && COMMON_CLK [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=y] This is needed because 'select' does not follow dependency chains. Fixes: a6f933f63f2f ("ASoC: qcom: apq8096: Add db820c machine driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- Found in linux-next but applies to mainline. sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20201016.orig/sound/soc/qcom/Kconfig +++ linux-next-20201016/sound/soc/qcom/Kconfig @@ -106,6 +106,7 @@ config SND_SOC_QDSP6 config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR + depends on COMMON_CLK select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help

2 2

[PATCH 0/2] Add rt1015 support to CML boards
by Brent Lu 30 Oct '20

30 Oct '20

First patch adds tdm 4-slot 100fs DAI setting to avoid jitter of using 64fs on CML boards. Second patch is a DMI quirk for HP Dooly. Brent Lu (2): ASoC: intel: sof_rt5682: Add support for cml_rt1015_rt5682 ASoC: intel: sof_rt5682: Add quirk for Dooly sound/soc/intel/boards/sof_rt5682.c | 61 +++++++++++++++++-- .../intel/common/soc-acpi-intel-cml-match.c | 13 ++++ 2 files changed, 69 insertions(+), 5 deletions(-) -- 2.17.1

4 6

[PATCH v3] soundwire: SDCA: add helper macro to access controls
by Bard Liao 30 Oct '20

30 Oct '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hierarchical encoding to interface with Class-defined capabilities. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. A block of 64 MBytes of register addresses are allocated to SDCA controls, starting at address 0x40000000. The 26 LSBs which identify individual controls are set based on the following variables: - Function Number. An SCDA device can be split in up to 8 independent Functions. Each of these Functions is described in the SDCA specification, e.g. Smart Amplifier, Smart Microphone, Simple Microphone, Jack codec, HID, etc. - Entity Number. Within each Function, an Entity is an identifiable block. Up to 127 Entities are connected in a pre-defined graph (similar to USB), with Entity0 reserved for Function-level configurations. In contrast to USB, the SDCA spec pre-defines Function Types, topologies, and allowed options, i.e. the degree of freedom is not unlimited to limit the possibility of errors in descriptors leading to software quirks. - Control Selector. Within each Entity, the SDCA specification defines 48 controls such as Mute, Gain, AGC, etc, and 16 implementation defined ones. Some Control Selectors might be used for low-level platform setup, and other exposed to applications and users. Note that the same Control Selector capability, e.g. Latency control, might be located at different offsets in different entities, the Control Selector mapping is Entity-specific. - Control Number. Some Control Selectors allow channel-specific values to be set, with up to 64 channels allowed. This is mostly used for volume control. - Current/Next values. Some Control Selectors are 'Dual-Ranked'. Software may either update the Current value directly for immediate effect. Alternatively, software may write into the 'Next' values and update the SoundWire 1.2 'Commit Groups' register to copy 'Next' values into 'Current' ones in a synchronized manner. This is different from bank switching which is typically used to change the bus configuration only. - MBQ. the Multi-Byte Quantity bit is used to provide atomic updates when accessing more that one byte, for example a 16-bit volume control would be updated consistently, the intermediate values mixing old MSB with new LSB are not applied. These 6 parameters are used to build a 32-bit address to access the desired Controls. Because of address range, paging is required, but the most often used parameter values are placed in the lower 16 bits of the address. This helps to keep the paging registers constant while updating Controls for a specific Device/Function. Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changelog: v2: - add SDW_SDCA_MBQ_CTL v3: - add SDW_SDCA_NEXT_CTL --- include/linux/soundwire/sdw_registers.h | 32 +++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index f420e8059779..e14dff9a9c7f 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -298,4 +298,36 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* + * v1.2 device - SDCA address mapping + * + * Spec definition + * Bits Contents + * 31 0 (required by addressing range) + * 30:26 0b10000 (Control Prefix) + * 25 0 (Reserved) + * 24:22 Function Number [2:0] + * 21 Entity[6] + * 20:19 Control Selector[5:4] + * 18 0 (Reserved) + * 17:15 Control Number[5:3] + * 14 Next + * 13 MBQ + * 12:7 Entity[5:0] + * 6:3 Control Selector[3:0] + * 2:0 Control Number[2:0] + */ + +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + +#define SDW_SDCA_MBQ_CTL(reg) ((reg) | BIT(13)) +#define SDW_SDCA_NEXT_CTL(reg) ((reg) | BIT(14)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

5 6

[PATCH] ALSA: ac97: fix reference leak in ac97_bus_remove
by Zhang Qilong 30 Oct '20

30 Oct '20

pm_runtime_get_sync() will increase pm usage counter even it failed. We should call pm_runtime_put_noidle to decrease it to prevent reference leak. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/ac97/bus.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7985dd8198b6..2c4779d851ad 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -521,8 +521,11 @@ static int ac97_bus_remove(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + dev_err(dev, "Failed to enable device: %d\n", ret); + pm_runtime_put_noidle(dev); return ret; + } ret = adrv->remove(adev); pm_runtime_put_noidle(dev); -- 2.17.1

1 0

[PATCH v4 00/15] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 30 Oct '20

30 Oct '20

Summary of changes: * Support multiple instances of a component. For example there can be multiple I2S devices which can use the same component driver. * Support open platforms with empty Codec endpoint. Customers can plug their own HW and can populate codec endpoint. * In a component model there can be many components which can be connected togethe. In such cases Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add Tegra audio graph driver which is based on generic audio graph driver and specific customizations are done in Tegra driver. * This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Changelog ========= v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (15): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: audio-graph: Expose new members for asoc_simple_priv ASoC: audio-graph: Expose helpers from audio graph Documentation: of: Convert graph bindings to json-schema ASoC: dt-bindings: audio-graph: Convert bindings to json-schema ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 Documentation/devicetree/bindings/graph.txt | 128 ----- Documentation/devicetree/bindings/graph.yaml | 170 +++++++ .../devicetree/bindings/sound/audio-graph-card.txt | 337 ------------- .../bindings/sound/audio-graph-card.yaml | 548 +++++++++++++++++++++ .../sound/nvidia,tegra-audio-graph-card.yaml | 158 ++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 7 + .../bindings/sound/nvidia,tegra210-admaif.yaml | 7 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 7 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 7 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 7 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 ++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++ arch/arm64/configs/defconfig | 1 + include/sound/graph_card.h | 16 + include/sound/simple_card_utils.h | 3 + include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 175 ++++--- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 255 ++++++++++ 23 files changed, 1882 insertions(+), 523 deletions(-) delete mode 100644 Documentation/devicetree/bindings/graph.txt create mode 100644 Documentation/devicetree/bindings/graph.yaml delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 include/sound/graph_card.h create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

5 30

[PATCH v3 1/2] ASoC: tegra20-spdif: remove "default m"
by Michał Mirosław 30 Oct '20

30 Oct '20

Make tegra20-spdif default to N as all other drivers do. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Fixes: 774fec338bfc ("ASoC: Tegra: Implement SPDIF CPU DAI") --- v3: split-off the defconfig changes v2: add the symbol to defconfig as suggested by Thierry Reding --- sound/soc/tegra/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 3d91bd3e59cd..a62cc87551ac 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -39,7 +39,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support -- 2.20.1

2 2

[PATCH v9 00/14] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 30 Oct '20

30 Oct '20

Hi, This series add H6 I2S support and the I2S node missing to support HDMI audio in different Allwinner SoC. As we first use some TDM property to make the I2S working we the simple soundcard. We have now drop this simple sound card and a proper dedicated soundcard will be introduce later. This make the title of this series wrong now :/. Regards, Clement Change since v8: - move the comment near the function prototype - collect Maxime Ripard tags Change since v7: - rebase on next-20201026 - comment about slots and slot_width Change since v6: - move set_channel_cfg() in first position - convert return value to decimal Change since v5: - Drop HDMI simple soundcard - Collect Chen-Yu Tsai tags - Configure channels from 9 to 15. - Remove DMA RX for H3/H5 - Fix Documentation for H3/H5 Change since v4: - add more comment on get_wss() and set_channel_cfg() patch - merge soundcard and DAI HDMI patches Change since v3: - add Samuel Holland patch to reconfigure FIFO_TX_REG when suspend is enabled - readd inversion to H6 LRCK sun50i_h6_i2s_set_soc_fmt() - Fix get_wss() for sun4i - Add a commit to fix checkpatch warning Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (6): ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition arm64: defconfig: Enable Allwinner i2s driver dt-bindings: sound: sun4i-i2s: Document H3 with missing RX channel possibility Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add I2S1 node Marcus Cooper (4): ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm64: dts: allwinner: a64: Add I2S2 node arm: dts: sunxi: h3/h5: Add I2S2 node Samuel Holland (1): ASoC: sun4i-i2s: Fix setting of FIFO modes .../sound/allwinner,sun4i-a10-i2s.yaml | 6 +- arch/arm/boot/dts/sunxi-h3-h5.dtsi | 13 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 14 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 13 + arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 384 +++++++++++++++--- 6 files changed, 376 insertions(+), 55 deletions(-) -- 2.25.1

4 19

[PATCH] ASoC: nau8315: add codec driver
by David Lin 30 Oct '20

30 Oct '20

Add codec driver for Nuvoton NAU8315. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8315.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8315.c | 166 ++++++++++++++++++ 4 files changed, 191 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt create mode 100644 sound/soc/codecs/nau8315.c diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000000..6eaec46f384c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5791b7056af6..2b6eceb3573c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MT6358 imply SND_SOC_MT6359 imply SND_SOC_MT6660 + imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 imply SND_SOC_NAU8822 @@ -1760,6 +1761,10 @@ config SND_SOC_MT6660 Select N if you don't have MT6660 on board. Select M to build this as module. +config SND_SOC_NAU8315 + tristate "Nuvoton Technology Corporation NAU8315 CODEC" + depends on GPIOLIB + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 11ce98c25d6c..144d9256d4d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,6 +129,7 @@ snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o snd-soc-mt6359-objs := mt6359.o snd-soc-mt6660-objs := mt6660.o +snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -438,6 +439,7 @@ obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o +obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(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 diff --git a/sound/soc/codecs/nau8315.c b/sound/soc/codecs/nau8315.c new file mode 100644 index 000000000000..e6bc5c0a5036 --- /dev/null +++ b/sound/soc/codecs/nau8315.c @@ -0,0 +1,166 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8315.c -- NAU8315 ALSA SoC Audio Amplifier Driver +// +// Copyright 2020 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0(a)nuvoton.com> +// +// Based on MAX98357A.c + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> + +struct nau8315_priv { + struct gpio_desc *enable; + int enpin_switch; +}; + +static int nau8315_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (!nau8315->enable) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (nau8315->enpin_switch) { + gpiod_set_value(nau8315->enable, 1); + dev_dbg(component->dev, "set enable to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(nau8315->enable, 0); + dev_dbg(component->dev, "set enable to 0"); + break; + } + + return 0; +} + +static int nau8315_enpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + nau8315->enpin_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + nau8315->enpin_switch = 0; + + return 0; +} + +static const struct snd_soc_dapm_widget nau8315_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("EN_Pin", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8315_enpin_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route nau8315_dapm_routes[] = { + {"EN_Pin", NULL, "HiFi Playback"}, + {"Speaker", NULL, "EN_Pin"}, +}; + +static const struct snd_soc_component_driver nau8315_component_driver = { + .dapm_widgets = nau8315_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8315_dapm_widgets), + .dapm_routes = nau8315_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8315_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops nau8315_dai_ops = { + .trigger = nau8315_daiops_trigger, +}; + +#define NAU8315_RATES SNDRV_PCM_RATE_8000_96000 +#define NAU8315_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver nau8315_dai_driver = { + .name = "nau8315-hifi", + .playback = { + .stream_name = "HiFi Playback", + .formats = NAU8315_FORMATS, + .rates = NAU8315_RATES, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &nau8315_dai_ops, +}; + +static int nau8315_platform_probe(struct platform_device *pdev) +{ + struct nau8315_priv *nau8315; + + nau8315 = devm_kzalloc(&pdev->dev, sizeof(*nau8315), GFP_KERNEL); + if (!nau8315) + return -ENOMEM; + + nau8315->enable = devm_gpiod_get_optional(&pdev->dev, + "enable", GPIOD_OUT_LOW); + if (IS_ERR(nau8315->enable)) + return PTR_ERR(nau8315->enable); + + dev_set_drvdata(&pdev->dev, nau8315); + + return devm_snd_soc_register_component(&pdev->dev, + &nau8315_component_driver, + &nau8315_dai_driver, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id nau8315_device_id[] = { + { .compatible = "nuvoton,nau8315" }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8315_device_id); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8315_acpi_match[] = { + { "NVTN2010", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8315_acpi_match); +#endif + +static struct platform_driver nau8315_platform_driver = { + .driver = { + .name = "nau8315", + .of_match_table = of_match_ptr(nau8315_device_id), + .acpi_match_table = ACPI_PTR(nau8315_acpi_match), + }, + .probe = nau8315_platform_probe, +}; +module_platform_driver(nau8315_platform_driver); + +MODULE_DESCRIPTION("ASoC NAU8315 Mono Class-D Amplifier Driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1 =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

3 3

[PATCH v2 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 30 Oct '20

30 Oct '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - fix indentation issue - remove nodename .../bindings/sound/fsl,aud2htx.yaml | 64 +++++++++++++++++++ 1 file changed, 64 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..6d9ba2946bfb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

3 3

[PATCH v2 0/2] ASoC: soc-compress: tidyup STREAM vs COMPRESS
by Kuninori Morimoto 30 Oct '20

30 Oct '20

Hi Mark These are v2 of tidyup patch-set for SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx. soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx, but mixed use. This is confusable, but no problem. Because these are defined as UAPI and are using same value. This patch-set make sure these are same value. Kuninori Morimoto (2): ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same sound/soc/soc-compress.c | 63 +++++++++++++++------------------------- 1 file changed, 23 insertions(+), 40 deletions(-) -- 2.25.1

1 1

[PATCH 0/2] ASoC: soc-compress: tidyup STREAM vs COMPRESS
by Kuninori Morimoto 30 Oct '20

30 Oct '20

Hi Mark soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx. It needs to be converted, but some of them didn't. This is bug, but this bug does nothing because these are using same value. This patch-set cleanup these. Kuninori Morimoto (2): ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: add soc_compr_cstream_to_stream() sound/soc/soc-compress.c | 65 ++++++++++++++++------------------------ 1 file changed, 25 insertions(+), 40 deletions(-) -- 2.25.1

3 9

[PATCH v2] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 30 Oct '20

30 Oct '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..a071485383ed 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -708,11 +708,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -720,11 +733,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -767,6 +794,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

1 0

[PATCH 1/2] ASoC: qcom: dt-bindings: sm8250: update compatibles
by Srinivas Kandagatla 29 Oct '20

29 Oct '20

Update compatible string as board compatible and device compatible should not be same!. New compatible is now suffixed with -sndcard to be inline with other Qualcomm Sound cards. This also fixes the warnings/error reported by dt_binding_check. Fixes: 765c37598494 ("ASoC: qcom: dt-bindings: Add SM8250 sound card bindings") Reported-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index b8f97fe6e92c..72ad9ab91832 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -16,10 +16,8 @@ description: properties: compatible: oneOf: - - const: qcom,qrb5165-rb5 - - items: - - const: qcom,sm8250 - - const: qcom,qrb5165-rb5 + - const: qcom,sm8250-sndcard + - const: qcom,qrb5165-rb5-sndcard audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array @@ -83,7 +81,7 @@ examples: #include <dt-bindings/sound/qcom,q6afe.h> #include <dt-bindings/sound/qcom,q6asm.h> sound { - compatible = "qcom,qrb5165-rb5"; + compatible = "qcom,qrb5165-rb5-sndcard"; model = "Qualcomm-qrb5165-RB5-WSA8815-Speakers-DMIC0"; audio-routing = "SpkrLeft IN", "WSA_SPK1 OUT", "SpkrRight IN", "WSA_SPK2 OUT", -- 2.21.0

2 2

[PATCH 01/25] ALSA: core: pcm: remove unnecessary CONFIG_PM_SLEEP
by Coiby Xu 29 Oct '20

29 Oct '20

SET_SYSTEM_SLEEP_PM_OPS has already took good care of CONFIG_PM_CONFIG. Signed-off-by: Coiby Xu <coiby.xu(a)gmail.com> --- sound/core/pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index be5714f1bb58..5a281ac92958 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -599,7 +599,6 @@ static const struct attribute_group *pcm_dev_attr_groups[]; * PM callbacks: we need to deal only with suspend here, as the resume is * triggered either from user-space or the driver's resume callback */ -#ifdef CONFIG_PM_SLEEP static int do_pcm_suspend(struct device *dev) { struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); @@ -608,7 +607,6 @@ static int do_pcm_suspend(struct device *dev) snd_pcm_suspend_all(pstr->pcm); return 0; } -#endif static const struct dev_pm_ops pcm_dev_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(do_pcm_suspend, NULL) -- 2.28.0

4 29

[PATCH v2] Asoc: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions
by Srinivasa Rao Mandadapu 29 Oct '20

29 Oct '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Update SC7180 lpass_variant structure with proper I2S bitwidth field bit positions, as bitwidth denotes 0 to 1 bits, but previously used only 0 bit. Fixes: cba62c8b49be ("ASoC: qcom: Add support for SC7180 lpass variant") Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since v1: -- Commit message is changed sound/soc/qcom/lpass-sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index c6292f9e..bc998d5 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = { .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH v2] Asoc: qcom: lpass-cpu: Fix clock disable failure
by Srinivasa Rao Mandadapu 29 Oct '20

29 Oct '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Disable MI2S bit clock from PAUSE/STOP/SUSPEND usecase instead of shutdown time. Acheive this by invoking clk_disable API from cpu daiops trigger instead of cpu daiops shutdown. Change non-atomic API "clk_prepare_enable" to atomic API "clk_enable" in trigger, as trigger is being called from atomic context. Fixes: commit 7e6799d8f87d ("ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together") Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 78de888..6001049 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret); return ret; } + ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + return ret; + } return 0; } @@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); - clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]); + ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable_unprepare(drvdata->mi2s_osr_clk[id]); + clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); break; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH][next] ASoC: qcom: fix unsigned int bitwidth compared to less than zero
by Colin King 29 Oct '20

29 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> The check for an error return from the call to snd_pcm_format_width is never true as the unsigned int bitwidth can never be less than zero. Fix this by making bitwidth an int. Addresses-Coverity: ("Unsigned compared against 0") Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/lpass-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-hdmi.c b/sound/soc/qcom/lpass-hdmi.c index 172952d3a5d6..abfb8737a89f 100644 --- a/sound/soc/qcom/lpass-hdmi.c +++ b/sound/soc/qcom/lpass-hdmi.c @@ -24,7 +24,7 @@ static int lpass_hdmi_daiops_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(params); unsigned int channels = params_channels(params); unsigned int ret; - unsigned int bitwidth; + int bitwidth; unsigned int word_length; unsigned int ch_sts_buf0; unsigned int ch_sts_buf1; -- 2.27.0

2 1

[PATCH v1 0/2] ASoC: qcom: add support for QRB5165 RB5 machine
by Srinivas Kandagatla 29 Oct '20

29 Oct '20

This patchset adds support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Srinivas Kandagatla (2): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support .../bindings/sound/qcom,sm8250.yaml | 161 +++++++++++++ sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sm8250.c | 228 ++++++++++++++++++ 4 files changed, 402 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml create mode 100644 sound/soc/qcom/sm8250.c -- 2.21.0

3 7

ucm: return error if boot_list is empty
by GitHub pull_request - opened 29 Oct '20

29 Oct '20

alsa-project/alsa-lib pull request #94 was opened from jason77-wang: If bootlist is empty, that means there is no BootSequence defined in the ucm for this sound card, let set_boot_user() return error, then alsactl could have chance to use non-ucm ways to initialize the sound card. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/94 Patch URL : https://github.com/alsa-project/alsa-lib/pull/94.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix initsetting
by GitHub pull_request - opened 29 Oct '20

29 Oct '20

alsa-project/alsa-ucm-conf pull request #57 was opened from jason77-wang: This issue was found when we prepared ubuntu 20.10. When users install the OS to a machine with HDA codec and use sof driver, the audio output is muted after installation. And even we set the master playback switch to on, the output volume from speaker is too low. This issue could be easily reproduced by: sudo rm /var/lib/asla/* rm ~/.config/pulse/* (optional) sudo /usr/sbin/alsactl -E HOME=/run/alsa -E XDG_RUNTIME_DIR=/run/alsa/runtime restore Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/57 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/57.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 29 Oct '20

29 Oct '20

Hi Rob, Mark Cc: Sameer These are v4 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 165 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 623 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

2 5

[PATCH v3 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 29 Oct '20

29 Oct '20

Hi Rob, Mark Cc: Sameer These are v3 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 165 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 623 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

3 15

[PATCH] ASoC: qcom: sm8250: Fix array out of bounds access
by Srinivas Kandagatla 28 Oct '20

28 Oct '20

Static analysis Coverity had detected a potential array out-of-bounds write issue due to the fact that MAX AFE port Id was set to 16 instead of using AFE_PORT_MAX macro. Fix this by properly using AFE_PORT_MAX macro. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Reported-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 7d43de6d909f..52c40512102f 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -13,12 +13,11 @@ #define DRIVER_NAME "sm8250" #define MI2S_BCLK_RATE 1536000 -#define MAX_SDW_STREAMS 16 struct sm8250_snd_data { - bool stream_prepared[MAX_SDW_STREAMS]; + bool stream_prepared[AFE_PORT_MAX]; struct snd_soc_card *card; - struct sdw_stream_runtime *sruntime[MAX_SDW_STREAMS]; + struct sdw_stream_runtime *sruntime[AFE_PORT_MAX]; }; static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, -- 2.21.0

3 2

[PATCH 1/2] ASoC: codecs: wcd934x: Set digital gain range correctly
by Srinivas Kandagatla 28 Oct '20

28 Oct '20

digital gain range is -84dB min to 40dB max, however this was not correctly specified in the range. Fix this by with correct range! Fixes: 1cde8b822332 ("ASoC: wcd934x: add basic controls") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 35697b072367..40f682f5dab8 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl { struct soc_bytes_ext bytes_ext; }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); -- 2.21.0

2 2

[PATCH v3 0/3] HDA controller PM and codec PM cleanups
by Kai-Heng Feng 28 Oct '20

28 Oct '20

While working on the issue "ALSA: hda: fix jack detection with Realtek codecs when in D3", I've found using pm_runtime_force_{suspend,resume}() have surprising behavior, specifically, pm_runtime_need_not_resume() uses pm_runtime_need_not_resume() to avoid calling resume callback, so jackpoll was used to really power up the codec. We can use direct-complete to do the keep the codec suspended throughout the system PM flow, namely, keep the codec suspended all the way, unless the codec needs to be woken up after resume. For HDA controller, PCI core may enable direct-complete for it if conditions are met. So make runtime and system PM distinctive to always apply correct wake up setting. At least point, hopefully all runtime PM issues are solved, let's enable runtime PM by default again. v3: - Drop "ALSA: hda: Stop mangling PCI IRQ" Kai-Heng Feng (3): ALSA: hda: Refactor codec PM to use direct-complete optimization ALSA: hda: Separate runtime and system suspend ALSA: hda: Reinstate runtime_allow() for all hda controllers sound/pci/hda/hda_codec.c | 45 +++++++++++++++--------- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 63 +++++++++++++++++++--------------- 3 files changed, 66 insertions(+), 45 deletions(-) -- 2.17.1

2 4

re: ASoC: qcom: sm8250: add sound card qrb5165-rb5 support
by Colin Ian King 28 Oct '20

28 Oct '20

Hi, Static analysis on linux-next with Coverity had detected a potential array out-of-bounds write issue in the following commit: commit aa2e2785545aab21b6cb2e23f111ae0751cbcca7 Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Mon Oct 26 17:09:47 2020 +0000 ASoC: qcom: sm8250: add sound card qrb5165-rb5 support The analysis is as follows: 139 static int sm8250_snd_hw_free(struct snd_pcm_substream *substream) 140 { 141 struct snd_soc_pcm_runtime *rtd = substream->private_data; 142 struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card); 143 struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); 144 struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; 145 1. Switch case value 105. 146 switch (cpu_dai->id) { 2. equality_cond: Jumping to case 105. 147 case WSA_CODEC_DMA_RX_0: 148 case WSA_CODEC_DMA_RX_1: Out-of-bounds write (OVERRUN) 3. Condition sruntime, taking true branch. 4. Condition data->stream_prepared[cpu_dai->id], taking true branch. 149 if (sruntime && data->stream_prepared[cpu_dai->id]) { 150 sdw_disable_stream(sruntime); 151 sdw_deprepare_stream(sruntime); Out-of-bounds write (OVERRUN) 5. overrun-local: Overrunning array data->stream_prepared of 16 bytes at byte offset 105 using index cpu_dai->id (which evaluates to 105). 152 data->stream_prepared[cpu_dai->id] = false; 153 } 154 break; 155 default: 156 break; 157 } 158 159 return 0; 160 } So cpu_dia->id is 105 in this case statement, and yet data->steam_prepared is an array of 16 elements, so this looks suspect. Colin

2 1

HP AIO Headset Mic issue
by Kailang 28 Oct '20

28 Oct '20

Hi Takashi, HP AIO had headset Mic can't be detected by boot time. Attach patch will fix it. Many Thanks. Kailang

2 1

[PATCH 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 28 Oct '20

28 Oct '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../bindings/sound/fsl,aud2htx.yaml | 67 +++++++++++++++++++ 1 file changed, 67 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..18548d0889a8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + $nodename: + pattern: "^aud2htx@.*" + + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

4 5

Out of bounds array reference aplay.c playbackv_go function
by GitHub issues - opened 27 Oct '20

27 Oct '20

alsa-project/alsa-utils issue #70 was opened from ygator: I was looking through the aplay source file and came across an out of bounds reference. The reason I was looking through the code is that I am using an i2s amp and there is this code: `/usr/bin/aplay -D default -t raw -r 44100 -c 2 -f S16_LE /dev/zero /dev/zero` The reason for it is: > We've added an extra helper systemd script that will play quiet audio when the I2S peripheral isn't in use. This removes popping when playback starts or stops. It uses a tiny amount of CPU time (on a Pi Zero, 5%, on a Pi 2 or 3 its negligible). I thought there must be a way to just open the device and not have to use any CPU time actually playing nothing. If anyone knows if this is possible then let me know. Thanks. Any way back to the out of bounds. In the function playbackv_go function of aplay.c there is a call to perror(names[channel]). However, instead of channel being equal to zero it is equal to channels which is 1 + the number of names being passed to the function. > while (count > 0 && !in_aborting) { size_t c = 0; size_t expected = count / channels; if (expected > vsize) expected = vsize; do { **channel = 0; // Code to fix out of bound** r = safe_read(fds[0], bufs[0], expected); if (r < 0) { **perror(names[channel]); // Out of bounds reference** prg_exit(EXIT_FAILURE); } for (channel = 1; channel < channels; ++channel) { if (safe_read(fds[channel], bufs[channel], r) != r) { perror(names[channel]); prg_exit(EXIT_FAILURE); } } if (r == 0) break; c += r; } while (c < expected); Issue URL : https://github.com/alsa-project/alsa-utils/issues/70 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] drm/vc4: hdmi: Avoid sleeping in atomic context
by Maxime Ripard 27 Oct '20

27 Oct '20

When running the trigger hook, ALSA by default will take a spinlock, and thus will run the trigger hook in atomic context. However, our HDMI driver will send the infoframes as part of the trigger hook, and part of that process is to wait for a bit to be cleared for up to 100ms. To be nicer to the system, that wait has some usleep_range that interact poorly with the atomic context. There's several ways we can fix this, but the more obvious one is to make ALSA take a mutex instead by setting the nonatomic flag on the DAI link. That doesn't work though, since now the cyclic callback installed by the dmaengine helpers in ALSA will take a mutex, while that callback is run by dmaengine's virt-chan code in a tasklet where sleeping is not allowed either. Given the delay we need to poll the bit for, changing the usleep_range for a udelay and keep running it from a context where interrupts are disabled is not really a good option either. However, we can move the infoframe setup code in the hw_params hook, like is usually done in other HDMI controllers, that isn't protected by a spinlock and thus where we can sleep. Infoframes will be sent on a regular basis anyway, and since hw_params is where the audio parameters that end up in the infoframes are setup, this also makes a bit more sense. Fixes: bb7d78568814 ("drm/vc4: Add HDMI audio support") Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- drivers/gpu/drm/vc4/vc4_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/gpu/drm/vc4/vc4_hdmi.c b/drivers/gpu/drm/vc4/vc4_hdmi.c index 74da7c00ecd0..ec3ba3ecd32a 100644 --- a/drivers/gpu/drm/vc4/vc4_hdmi.c +++ b/drivers/gpu/drm/vc4/vc4_hdmi.c @@ -1077,6 +1077,7 @@ static int vc4_hdmi_audio_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct vc4_hdmi *vc4_hdmi = dai_to_hdmi(dai); + struct drm_encoder *encoder = &vc4_hdmi->encoder.base.base; struct device *dev = &vc4_hdmi->pdev->dev; u32 audio_packet_config, channel_mask; u32 channel_map; @@ -1136,6 +1137,8 @@ static int vc4_hdmi_audio_hw_params(struct snd_pcm_substream *substream, HDMI_WRITE(HDMI_AUDIO_PACKET_CONFIG, audio_packet_config); vc4_hdmi_set_n_cts(vc4_hdmi); + vc4_hdmi_set_audio_infoframe(encoder); + return 0; } @@ -1143,11 +1146,9 @@ static int vc4_hdmi_audio_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct vc4_hdmi *vc4_hdmi = dai_to_hdmi(dai); - struct drm_encoder *encoder = &vc4_hdmi->encoder.base.base; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - vc4_hdmi_set_audio_infoframe(encoder); vc4_hdmi->audio.streaming = true; if (vc4_hdmi->variant->phy_rng_enable) -- 2.26.2

3 3

dapm, device tree, clock control - questions
by Eliot Blennerhassett 27 Oct '20

27 Oct '20

Greetings, First the basic questions, so you can quickly stop reading if this is not your thing. Background details follow. Specific answers, or pointers to relevant reading both appreciated. 1) How do I force a DAI clock to keep running even when no audio is streaming? Can I do this with DT, or some runtime switch? OR how do I disable dapm so everything is powered on all the time? 2) Two devices: one DAC, one dumb digital audio transmitter both connected to same serial audio data - how to model the second one in DT (or not)? it has some settings needed after clock starts, must keep clock running. I have host CPU imx8mm DAI is sound/soc/fsl/fsl_sai.c The codec is sound/soc/codecs/ak4458.c The digital audio transmitter is CS8406 (no driver yet?). Both the above are connected to the same DAI signals (MCLK, TXC, TXFS, TXD) CS8406 needs MCLK before it can be configured, and loses config if clock stops. Currently it is configured via I2C from userspace, but this is completely unaware of stream stop/start. Is there a way to hook into stream state events so that this setup can be done after the MCLK starts before audio starts? I desire for no discontinuity in analog or digital outputs as streams stop and start. I have read Documentation/sound/soc/dapm.rst and its brothers. Documentation/devicetree/* is there elsewhere I should read to get a newbies intro to this stuff? It must be about 8 years since I did any ALSA development, and then I wasn't in the ASoC space, so quite a learning curve here. many thanks -- Eliot

1 1

[PATCH v2] ASoC: qcom: qdsp6: make use of devm_of_platform_populate
by Srinivas Kandagatla 27 Oct '20

27 Oct '20

make use of devm_of_platform_populate to remove some redundant code! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: rebased on top of sound-next sound/soc/qcom/qdsp6/q6adm.c | 10 +--------- sound/soc/qcom/qdsp6/q6afe.c | 10 +--------- sound/soc/qcom/qdsp6/q6asm.c | 10 +--------- 3 files changed, 3 insertions(+), 27 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 72f29720398c..1855b805eba2 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -601,14 +601,7 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - return of_platform_populate(dev->of_node, NULL, NULL, dev); -} - -static int q6adm_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; + return devm_of_platform_populate(dev); } #ifdef CONFIG_OF @@ -621,7 +614,6 @@ MODULE_DEVICE_TABLE(of, q6adm_device_id); static struct apr_driver qcom_q6adm_driver = { .probe = q6adm_probe, - .remove = q6adm_remove, .callback = q6adm_callback, .driver = { .name = "qcom-q6adm", diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 0ca1e4aae518..daa58b5f941e 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1740,14 +1740,7 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - return of_platform_populate(dev->of_node, NULL, NULL, dev); -} - -static int q6afe_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; + return devm_of_platform_populate(dev); } #ifdef CONFIG_OF @@ -1760,7 +1753,6 @@ MODULE_DEVICE_TABLE(of, q6afe_device_id); static struct apr_driver qcom_q6afe_driver = { .probe = q6afe_probe, - .remove = q6afe_remove, .callback = q6afe_callback, .driver = { .name = "qcom-q6afe", diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index c547c560cb24..a6618efe22f2 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1727,14 +1727,7 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - return of_platform_populate(dev->of_node, NULL, NULL, dev); -} - -static int q6asm_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; + return devm_of_platform_populate(dev); } #ifdef CONFIG_OF @@ -1747,7 +1740,6 @@ MODULE_DEVICE_TABLE(of, q6asm_device_id); static struct apr_driver qcom_q6asm_driver = { .probe = q6asm_probe, - .remove = q6asm_remove, .callback = q6asm_srvc_callback, .driver = { .name = "qcom-q6asm", -- 2.21.0

2 1

[PATCH] ASoC: qcom: qdsp6: make use of devm_of_platform_populate
by Srinivas Kandagatla 27 Oct '20

27 Oct '20

make use of devm_of_platform_populate to remove some redundant code! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6adm.c | 10 +--------- sound/soc/qcom/qdsp6/q6afe.c | 10 +--------- sound/soc/qcom/qdsp6/q6asm.c | 9 +-------- 3 files changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 2f3ea6beb066..e4b3d566447a 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -601,14 +601,7 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - return of_platform_populate(dev->of_node, NULL, NULL, dev); -} - -static int q6adm_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; + return devm_of_platform_populate(dev); } static const struct of_device_id q6adm_device_id[] = { @@ -619,7 +612,6 @@ MODULE_DEVICE_TABLE(of, q6adm_device_id); static struct apr_driver qcom_q6adm_driver = { .probe = q6adm_probe, - .remove = q6adm_remove, .callback = q6adm_callback, .driver = { .name = "qcom-q6adm", diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 688878a002a4..2efde6bd2123 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -1740,14 +1740,7 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - return of_platform_populate(dev->of_node, NULL, NULL, dev); -} - -static int q6afe_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; + return devm_of_platform_populate(dev); } static const struct of_device_id q6afe_device_id[] = { @@ -1758,7 +1751,6 @@ MODULE_DEVICE_TABLE(of, q6afe_device_id); static struct apr_driver qcom_q6afe_driver = { .probe = q6afe_probe, - .remove = q6afe_remove, .callback = q6afe_callback, .driver = { .name = "qcom-q6afe", diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index d745a02fcd5f..0965c0d4ac2d 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1724,15 +1724,9 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - return of_platform_populate(dev->of_node, NULL, NULL, dev); + return devm_of_platform_populate(dev); } -static int q6asm_remove(struct apr_device *adev) -{ - of_platform_depopulate(&adev->dev); - - return 0; -} static const struct of_device_id q6asm_device_id[] = { { .compatible = "qcom,q6asm" }, {}, @@ -1741,7 +1735,6 @@ MODULE_DEVICE_TABLE(of, q6asm_device_id); static struct apr_driver qcom_q6asm_driver = { .probe = q6asm_probe, - .remove = q6asm_remove, .callback = q6asm_srvc_callback, .driver = { .name = "qcom-q6asm", -- 2.21.0

2 2

[PATCH v12 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 27 Oct '20

27 Oct '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Changes from v9 to v10 - Documentation: - Let compatible string be more specific for board configuration to allow for future changes. - Machine driver: - Fixed unused include and macro. - Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h. - Let sound card be dynamically allocated. - Change compatible string accordingly. Changes from v10 to v11 - Machine driver: - Use temporary macro LPASS_DP_RX for future change in sc7180-lpass.h. Changes from v11 to v12 - Documentation: - Change the file and title name for new compatible string google,sc7180-trogdor. - Change the example of model name. - Machine driver: - Use the definitaion of index LPASS_DP_RX in sc7180-lpass.h. - Fix for compatible string. - Replace a comma with semicolon. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: google: dt-bindings: Add sc7180-trogdor machine bindings .../bindings/sound/google,sc7180-trogdor.yaml | 130 +++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 266 ++++++++++++++++++ 4 files changed, 410 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.29.0.rc2.309.g374f81d7ae-goog

3 5

how to properly release all memory after use?
by GitHub issues - edited 27 Oct '20

27 Oct '20

alsa-project/alsa-lib issue #93 was edited from xatian: ```C++ int main () { snd_pcm_t* handle; int err; err = ::snd_pcm_open (&handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); ::printf ("snd_pcm_open: %d\n", err); err = ::snd_pcm_close (handle); ::printf ("snd_pcm_close: %d\n", err); ::snd_config_update_free_global (); return 0; } ``` ``` ==51583== Memcheck, a memory error detector ==51583== Copyright (C) 2002-2017, and GNU GPL'd, by Julian Seward et al. ==51583== Using Valgrind-3.16.1 and LibVEX; rerun with -h for copyright info ==51583== Command: ./alsa-leak ==51583== snd_pcm_open: 0 snd_pcm_close: 0 ==51583== ==51583== HEAP SUMMARY: ==51583== in use at exit: 64,140 bytes in 140 blocks ==51583== total heap usage: 2,937 allocs, 2,797 frees, 635,030 bytes allocated ==51583== ==51583== 4 bytes in 1 blocks are definitely lost in loss record 3 of 71 ==51583== at 0x483AB65: calloc (vg_replace_malloc.c:760) ==51583== by 0x58E0A3E: pa_xmalloc0 (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x58B5F9E: pa_context_new_with_proplist (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x4848216: ??? ==51583== by 0x4904ABD: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4904F24: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905849: snd_config_searcha_hooks (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905940: snd_config_searchva_hooks (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905A42: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x49042D8: snd_config_search_definition (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4920255: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4923227: snd_pcm_open (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== ==51583== 4 bytes in 1 blocks are definitely lost in loss record 4 of 71 ==51583== at 0x483AB65: calloc (vg_replace_malloc.c:760) ==51583== by 0x58E0A3E: pa_xmalloc0 (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x58B5F9E: pa_context_new_with_proplist (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x5892577: ??? ==51583== by 0x5891F42: ??? ==51583== by 0x491FC84: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x49202E6: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4923227: snd_pcm_open (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x404A5B: main (src/main.cpp:15) ==51583== ==51583== LEAK SUMMARY: ==51583== definitely lost: 8 bytes in 2 blocks ==51583== indirectly lost: 0 bytes in 0 blocks ==51583== possibly lost: 0 bytes in 0 blocks ==51583== still reachable: 64,132 bytes in 138 blocks ==51583== suppressed: 0 bytes in 0 blocks ==51583== Reachable blocks (those to which a pointer was found) are not shown. ==51583== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==51583== ==51583== For lists of detected and suppressed errors, rerun with: -s ==51583== ERROR SUMMARY: 2 errors from 2 contexts (suppressed: 0 from 0) ``` Did I do something wrong? What makes matters worse is that if I duplicate the code more leaks get reported so this is not just some global state that gets left behind. I am using libasound2 1.2.3.2-1 amd64 Thank you! Issue URL : https://github.com/alsa-project/alsa-lib/issues/93 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

how to properly release all memory after use?
by GitHub issues - opened 27 Oct '20

27 Oct '20

alsa-project/alsa-lib issue #93 was opened from xatian: ```C++ int main () { snd_pcm_t* handle; int err; err = ::snd_pcm_open (&handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); ::printf ("snd_pcm_open: %d\n", err); err = ::snd_pcm_close (handle); ::printf ("snd_pcm_close: %d\n", err); ::snd_config_update_free_global (); return 0; } ``` ``` ==51583== Memcheck, a memory error detector ==51583== Copyright (C) 2002-2017, and GNU GPL'd, by Julian Seward et al. ==51583== Using Valgrind-3.16.1 and LibVEX; rerun with -h for copyright info ==51583== Command: ./alsa-leak ==51583== snd_pcm_open: 0 snd_pcm_close: 0 ==51583== ==51583== HEAP SUMMARY: ==51583== in use at exit: 64,140 bytes in 140 blocks ==51583== total heap usage: 2,937 allocs, 2,797 frees, 635,030 bytes allocated ==51583== ==51583== 4 bytes in 1 blocks are definitely lost in loss record 3 of 71 ==51583== at 0x483AB65: calloc (vg_replace_malloc.c:760) ==51583== by 0x58E0A3E: pa_xmalloc0 (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x58B5F9E: pa_context_new_with_proplist (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x4848216: ??? ==51583== by 0x4904ABD: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4904F24: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905849: snd_config_searcha_hooks (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905940: snd_config_searchva_hooks (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4905A42: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x49042D8: snd_config_search_definition (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4920255: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4923227: snd_pcm_open (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== ==51583== 4 bytes in 1 blocks are definitely lost in loss record 4 of 71 ==51583== at 0x483AB65: calloc (vg_replace_malloc.c:760) ==51583== by 0x58E0A3E: pa_xmalloc0 (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x58B5F9E: pa_context_new_with_proplist (in /usr/lib/x86_64-linux-gnu/libpulse.so.0.21.1) ==51583== by 0x5892577: ??? ==51583== by 0x5891F42: ??? ==51583== by 0x491FC84: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x49202E6: ??? (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x4923227: snd_pcm_open (in /usr/lib/x86_64-linux-gnu/libasound.so.2.0.0) ==51583== by 0x404A5B: main (src/main.cpp:15) ==51583== ==51583== LEAK SUMMARY: ==51583== definitely lost: 8 bytes in 2 blocks ==51583== indirectly lost: 0 bytes in 0 blocks ==51583== possibly lost: 0 bytes in 0 blocks ==51583== still reachable: 64,132 bytes in 138 blocks ==51583== suppressed: 0 bytes in 0 blocks ==51583== Reachable blocks (those to which a pointer was found) are not shown. ==51583== To see them, rerun with: --leak-check=full --show-leak-kinds=all ==51583== ==51583== For lists of detected and suppressed errors, rerun with: -s ==51583== ERROR SUMMARY: 2 errors from 2 contexts (suppressed: 0 from 0) ``` Did I do something wrong? What makes matters worse is that if I duplicate the code more leaks get reported so this is not just some global state that gets left behind. And what about the 64kB still reachable? I am using libasound2 1.2.3.2-1 amd64 Thank you! Issue URL : https://github.com/alsa-project/alsa-lib/issues/93 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v8 00/14] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 27 Oct '20

27 Oct '20

Hi, This is the same as v7 but rebased on next-20201026 and added a comment about slots and slot_width. A proper sound card will be introduced later. This was tested on H6 only. Regards, Clement Change since v7: - rebase on next-20201026 - comment about slots and slot_width Change since v6: - move set_channel_cfg() in first position - convert return value to decimal Change since v5: - Drop HDMI simple soundcard - Collect Chen-Yu Tsai tags - Configure channels from 9 to 15. - Remove DMA RX for H3/H5 - Fix Documentation for H3/H5 Change since v4: - add more comment on get_wss() and set_channel_cfg() patch - merge soundcard and DAI HDMI patches Change since v3: - add Samuel Holland patch to reconfigure FIFO_TX_REG when suspend is enabled - readd inversion to H6 LRCK sun50i_h6_i2s_set_soc_fmt() - Fix get_wss() for sun4i - Add a commit to fix checkpatch warning Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (6): ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition arm64: defconfig: Enable Allwinner i2s driver dt-bindings: sound: sun4i-i2s: Document H3 with missing RX channel possibility Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add I2S1 node Marcus Cooper (4): ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm64: dts: allwinner: a64: Add I2S2 node arm: dts: sunxi: h3/h5: Add I2S2 node Samuel Holland (1): ASoC: sun4i-i2s: Fix setting of FIFO modes .../sound/allwinner,sun4i-a10-i2s.yaml | 6 +- arch/arm/boot/dts/sunxi-h3-h5.dtsi | 13 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 14 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 13 + arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 385 +++++++++++++++--- 6 files changed, 376 insertions(+), 56 deletions(-) -- 2.25.1

2 19

[PATCH v1 0/6] ASoC: codecs: add support for LPASS Codec macros
by Srinivas Kandagatla 27 Oct '20

27 Oct '20

This patchset adds support for two Codec Macro blocks( WSA and VA) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD Codecs to provide headphone/ear/lineout etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. yaml bindings depend on gfm headers from "clk: qcom : add sm8250 LPASS GFM drivers" patches posted at https://www.mail-archive.com/linux-kernel@vger.kernel.org/msg2352255.html Thanks, srini Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes .../bindings/sound/qcom,lpass-va-macro.yaml | 68 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 70 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-va-macro.c | 1500 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2503 +++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 15 + 7 files changed, 4168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h -- 2.21.0

2 11

[PATCH] dt-bindings: More whitespace clean-ups in schema files
by Rob Herring 27 Oct '20

27 Oct '20

Clean-up incorrect indentation, extra spaces, and missing EOF newline in schema files. Most of the clean-ups are for list indentation which should always be 2 spaces more than the preceding keyword. Found with yamllint (now integrated into the checks). Cc: linux-arm-kernel(a)lists.infradead.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../display/bridge/toshiba,tc358775.yaml | 18 ++-- .../display/panel/ilitek,ili9881c.yaml | 5 +- .../devicetree/bindings/eeprom/at25.yaml | 6 +- .../bindings/gpio/kontron,sl28cpld-gpio.yaml | 4 +- .../devicetree/bindings/i2c/ingenic,i2c.yaml | 8 +- .../bindings/iio/adc/adi,ad7291.yaml | 3 +- .../bindings/iio/adc/adi,ad7768-1.yaml | 3 +- .../bindings/iio/adc/cosmic,10001-adc.yaml | 4 +- .../bindings/iio/adc/holt,hi8435.yaml | 2 +- .../interrupt-controller/ti,pruss-intc.yaml | 12 +-- .../devicetree/bindings/mfd/ene-kb3930.yaml | 2 +- .../devicetree/bindings/mmc/arasan,sdhci.yaml | 8 +- .../devicetree/bindings/mmc/sdhci-am654.yaml | 15 ++- .../pci/socionext,uniphier-pcie-ep.yaml | 18 ++-- .../phy/socionext,uniphier-ahci-phy.yaml | 6 +- .../devicetree/bindings/phy/ti,omap-usb2.yaml | 20 ++-- .../pinctrl/actions,s500-pinctrl.yaml | 102 +++++++++--------- .../bindings/pinctrl/pinctrl-mt8192.yaml | 2 +- .../pinctrl/qcom,msm8226-pinctrl.yaml | 6 +- .../pinctrl/toshiba,visconti-pinctrl.yaml | 24 ++--- .../bindings/power/reset/reboot-mode.yaml | 6 +- .../power/supply/ingenic,battery.yaml | 8 +- .../power/supply/summit,smb347-charger.yaml | 16 +-- .../bindings/riscv/sifive-l2-cache.yaml | 4 +- .../devicetree/bindings/rng/imx-rng.yaml | 6 +- .../bindings/serial/fsl-imx-uart.yaml | 34 +++--- .../bindings/sound/mchp,spdifrx.yaml | 4 +- .../bindings/sound/mchp,spdiftx.yaml | 4 +- .../bindings/sound/qcom,lpass-cpu.yaml | 40 +++---- .../devicetree/bindings/timer/arm,sp804.yaml | 12 +-- .../devicetree/bindings/usb/cdns,usb3.yaml | 4 +- .../devicetree/bindings/usb/ti,hd3ss3220.yaml | 2 +- .../devicetree/bindings/w1/fsl-imx-owire.yaml | 8 +- 33 files changed, 210 insertions(+), 206 deletions(-) diff --git a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358775.yaml b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358775.yaml index 31f085d8ab13..526bbd63924b 100644 --- a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358775.yaml +++ b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358775.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Toshiba TC358775 DSI to LVDS bridge bindings maintainers: - - Vinay Simha BN <simhavcs(a)gmail.com> + - Vinay Simha BN <simhavcs(a)gmail.com> description: | This binding supports DSI to LVDS bridge TC358775 @@ -29,7 +29,7 @@ properties: vdd-supply: maxItems: 1 - description: 1.2V LVDS Power Supply + description: 1.2V LVDS Power Supply vddio-supply: maxItems: 1 @@ -77,13 +77,13 @@ properties: - port@1 required: - - compatible - - reg - - vdd-supply - - vddio-supply - - stby-gpios - - reset-gpios - - ports + - compatible + - reg + - vdd-supply + - vddio-supply + - stby-gpios + - reset-gpios + - ports examples: - | diff --git a/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml b/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml index c60b3bd74337..b2fcec4f22fd 100644 --- a/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml +++ b/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml @@ -13,9 +13,8 @@ properties: compatible: items: - enum: - - bananapi,lhr050h41 - - feixin,k101-im2byl02 - + - bananapi,lhr050h41 + - feixin,k101-im2byl02 - const: ilitek,ili9881c backlight: true diff --git a/Documentation/devicetree/bindings/eeprom/at25.yaml b/Documentation/devicetree/bindings/eeprom/at25.yaml index 9810619a2b5c..744973637678 100644 --- a/Documentation/devicetree/bindings/eeprom/at25.yaml +++ b/Documentation/devicetree/bindings/eeprom/at25.yaml @@ -81,14 +81,14 @@ properties: at25,byte-len: $ref: /schemas/types.yaml#/definitions/uint32 description: - Total eeprom size in bytes. Deprecated, use "size" property instead. + Total eeprom size in bytes. Deprecated, use "size" property instead. deprecated: true at25,addr-mode: $ref: /schemas/types.yaml#/definitions/uint32 description: - Addr-mode flags, as defined in include/linux/spi/eeprom.h. - Deprecated, use "address-width" property instead. + Addr-mode flags, as defined in include/linux/spi/eeprom.h. + Deprecated, use "address-width" property instead. deprecated: true at25,page-size: diff --git a/Documentation/devicetree/bindings/gpio/kontron,sl28cpld-gpio.yaml b/Documentation/devicetree/bindings/gpio/kontron,sl28cpld-gpio.yaml index e2d2c10e536a..b032471831e7 100644 --- a/Documentation/devicetree/bindings/gpio/kontron,sl28cpld-gpio.yaml +++ b/Documentation/devicetree/bindings/gpio/kontron,sl28cpld-gpio.yaml @@ -43,8 +43,8 @@ properties: gpio-controller: true gpio-line-names: - minItems: 1 - maxItems: 8 + minItems: 1 + maxItems: 8 required: - compatible diff --git a/Documentation/devicetree/bindings/i2c/ingenic,i2c.yaml b/Documentation/devicetree/bindings/i2c/ingenic,i2c.yaml index 0e7b4b8a7e48..e1e65eb4f795 100644 --- a/Documentation/devicetree/bindings/i2c/ingenic,i2c.yaml +++ b/Documentation/devicetree/bindings/i2c/ingenic,i2c.yaml @@ -19,11 +19,11 @@ properties: compatible: oneOf: - enum: - - ingenic,jz4770-i2c - - ingenic,x1000-i2c + - ingenic,jz4770-i2c + - ingenic,x1000-i2c - items: - - const: ingenic,jz4780-i2c - - const: ingenic,jz4770-i2c + - const: ingenic,jz4780-i2c + - const: ingenic,jz4770-i2c reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/iio/adc/adi,ad7291.yaml b/Documentation/devicetree/bindings/iio/adc/adi,ad7291.yaml index 6feafb7e531e..930f9e3904d7 100644 --- a/Documentation/devicetree/bindings/iio/adc/adi,ad7291.yaml +++ b/Documentation/devicetree/bindings/iio/adc/adi,ad7291.yaml @@ -43,4 +43,5 @@ examples: vref-supply = <&adc_vref>; }; }; -... \ No newline at end of file +... + diff --git a/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml b/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml index d3733ad8785a..8f32800fe5b7 100644 --- a/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml +++ b/Documentation/devicetree/bindings/iio/adc/adi,ad7768-1.yaml @@ -46,7 +46,8 @@ properties: spi-max-frequency: true spi-cpol: true - spi-cpha : true + + spi-cpha: true "#io-channel-cells": const: 1 diff --git a/Documentation/devicetree/bindings/iio/adc/cosmic,10001-adc.yaml b/Documentation/devicetree/bindings/iio/adc/cosmic,10001-adc.yaml index 5d92b477e23f..4e695b97d015 100644 --- a/Documentation/devicetree/bindings/iio/adc/cosmic,10001-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/cosmic,10001-adc.yaml @@ -22,8 +22,8 @@ properties: adc-reserved-channels: $ref: /schemas/types.yaml#/definitions/uint32 description: - Bitmask of reserved channels, i.e. channels that cannot be - used by the OS. + Bitmask of reserved channels, i.e. channels that cannot be + used by the OS. clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/iio/adc/holt,hi8435.yaml b/Documentation/devicetree/bindings/iio/adc/holt,hi8435.yaml index 9514c3381c42..52490cbb0af0 100644 --- a/Documentation/devicetree/bindings/iio/adc/holt,hi8435.yaml +++ b/Documentation/devicetree/bindings/iio/adc/holt,hi8435.yaml @@ -21,7 +21,7 @@ properties: gpios: description: - GPIO used for controlling the reset pin + GPIO used for controlling the reset pin maxItems: 1 spi-max-frequency: true diff --git a/Documentation/devicetree/bindings/interrupt-controller/ti,pruss-intc.yaml b/Documentation/devicetree/bindings/interrupt-controller/ti,pruss-intc.yaml index bbf79d125675..1c4c009dedd0 100644 --- a/Documentation/devicetree/bindings/interrupt-controller/ti,pruss-intc.yaml +++ b/Documentation/devicetree/bindings/interrupt-controller/ti,pruss-intc.yaml @@ -94,12 +94,12 @@ properties: instances. required: - - compatible - - reg - - interrupts - - interrupt-names - - interrupt-controller - - "#interrupt-cells" + - compatible + - reg + - interrupts + - interrupt-names + - interrupt-controller + - "#interrupt-cells" additionalProperties: false diff --git a/Documentation/devicetree/bindings/mfd/ene-kb3930.yaml b/Documentation/devicetree/bindings/mfd/ene-kb3930.yaml index 074243c40891..08af356f5d27 100644 --- a/Documentation/devicetree/bindings/mfd/ene-kb3930.yaml +++ b/Documentation/devicetree/bindings/mfd/ene-kb3930.yaml @@ -17,7 +17,7 @@ properties: compatible: items: - enum: - - dell,wyse-ariel-ec # Dell Wyse Ariel board (3020) + - dell,wyse-ariel-ec # Dell Wyse Ariel board (3020) - const: ene,kb3930 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/mmc/arasan,sdhci.yaml b/Documentation/devicetree/bindings/mmc/arasan,sdhci.yaml index 58fe9d02a781..0753289fba84 100644 --- a/Documentation/devicetree/bindings/mmc/arasan,sdhci.yaml +++ b/Documentation/devicetree/bindings/mmc/arasan,sdhci.yaml @@ -32,11 +32,11 @@ allOf: clock-output-names: oneOf: - items: - - const: clk_out_sd0 - - const: clk_in_sd0 + - const: clk_out_sd0 + - const: clk_in_sd0 - items: - - const: clk_out_sd1 - - const: clk_in_sd1 + - const: clk_out_sd1 + - const: clk_in_sd1 properties: compatible: diff --git a/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml b/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml index ac79f3adf20b..2cd859066462 100644 --- a/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml +++ b/Documentation/devicetree/bindings/mmc/sdhci-am654.yaml @@ -3,7 +3,7 @@ %YAML 1.2 --- $id: "http://devicetree.org/schemas/mmc/sdhci-am654.yaml#" -$schema : "http://devicetree.org/meta-schemas/core.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" title: TI AM654 MMC Controller @@ -163,13 +163,12 @@ properties: ti,driver-strength-ohm: description: DLL drive strength in ohms $ref: "/schemas/types.yaml#/definitions/uint32" - oneOf: - - enum: - - 33 - - 40 - - 50 - - 66 - - 100 + enum: + - 33 + - 40 + - 50 + - 66 + - 100 ti,strobe-sel: description: strobe select delay for HS400 speed mode. diff --git a/Documentation/devicetree/bindings/pci/socionext,uniphier-pcie-ep.yaml b/Documentation/devicetree/bindings/pci/socionext,uniphier-pcie-ep.yaml index f4292d2c54e3..d6cf8a560ef0 100644 --- a/Documentation/devicetree/bindings/pci/socionext,uniphier-pcie-ep.yaml +++ b/Documentation/devicetree/bindings/pci/socionext,uniphier-pcie-ep.yaml @@ -29,16 +29,16 @@ properties: reg-names: oneOf: - items: - - const: dbi - - const: dbi2 - - const: link - - const: addr_space + - const: dbi + - const: dbi2 + - const: link + - const: addr_space - items: - - const: dbi - - const: dbi2 - - const: link - - const: addr_space - - const: atu + - const: dbi + - const: dbi2 + - const: link + - const: addr_space + - const: atu clocks: maxItems: 2 diff --git a/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml b/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml index bab2ff4d9dc9..34756347a14e 100644 --- a/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml +++ b/Documentation/devicetree/bindings/phy/socionext,uniphier-ahci-phy.yaml @@ -31,10 +31,10 @@ properties: clock-names: oneOf: - items: # for PXs2 - - const: link + - const: link - items: # for others - - const: link - - const: phy + - const: link + - const: phy resets: maxItems: 2 diff --git a/Documentation/devicetree/bindings/phy/ti,omap-usb2.yaml b/Documentation/devicetree/bindings/phy/ti,omap-usb2.yaml index 15207ca9548f..ab77a65c5c55 100644 --- a/Documentation/devicetree/bindings/phy/ti,omap-usb2.yaml +++ b/Documentation/devicetree/bindings/phy/ti,omap-usb2.yaml @@ -7,23 +7,23 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: OMAP USB2 PHY maintainers: - - Kishon Vijay Abraham I <kishon(a)ti.com> - - Roger Quadros <rogerq(a)ti.com> + - Kishon Vijay Abraham I <kishon(a)ti.com> + - Roger Quadros <rogerq(a)ti.com> properties: compatible: oneOf: - items: - - enum: - - ti,dra7x-usb2 - - ti,dra7x-usb2-phy2 - - ti,am654-usb2 - - enum: - - ti,omap-usb2 + - enum: + - ti,dra7x-usb2 + - ti,dra7x-usb2-phy2 + - ti,am654-usb2 + - enum: + - ti,omap-usb2 - items: - - const: ti,am437x-usb2 + - const: ti,am437x-usb2 - items: - - const: ti,omap-usb2 + - const: ti,omap-usb2 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml index 33391d30c00c..ccdd9e3820d7 100644 --- a/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/actions,s500-pinctrl.yaml @@ -76,22 +76,22 @@ patternProperties: items: oneOf: - enum: [lcd0_d18_mfp, rmii_crs_dv_mfp, rmii_txd0_mfp, - rmii_txd1_mfp, rmii_txen_mfp, rmii_rxen_mfp, rmii_rxd1_mfp, - rmii_rxd0_mfp, rmii_ref_clk_mfp, i2s_d0_mfp, i2s_pcm1_mfp, - i2s0_pcm0_mfp, i2s1_pcm0_mfp, i2s_d1_mfp, ks_in2_mfp, - ks_in1_mfp, ks_in0_mfp, ks_in3_mfp, ks_out0_mfp, - ks_out1_mfp, ks_out2_mfp, lvds_o_pn_mfp, dsi_dn0_mfp, - dsi_dp2_mfp, lcd0_d17_mfp, dsi_dp3_mfp, dsi_dn3_mfp, - dsi_dp0_mfp, lvds_ee_pn_mfp, spi0_i2c_pcm_mfp, - spi0_i2s_pcm_mfp, dsi_dnp1_cp_mfp, lvds_e_pn_mfp, - dsi_dn2_mfp, uart2_rtsb_mfp, uart2_ctsb_mfp, uart3_rtsb_mfp, - uart3_ctsb_mfp, sd0_d0_mfp, sd0_d1_mfp, sd0_d2_d3_mfp, - sd1_d0_d3_mfp, sd0_cmd_mfp, sd0_clk_mfp, sd1_cmd_mfp, - uart0_rx_mfp, clko_25m_mfp, csi_cn_cp_mfp, sens0_ckout_mfp, - uart0_tx_mfp, i2c0_mfp, csi_dn_dp_mfp, sen0_pclk_mfp, - pcm1_in_mfp, pcm1_clk_mfp, pcm1_sync_mfp, pcm1_out_mfp, - dnand_data_wr_mfp, dnand_acle_ce0_mfp, nand_ceb2_mfp, - nand_ceb3_mfp] + rmii_txd1_mfp, rmii_txen_mfp, rmii_rxen_mfp, rmii_rxd1_mfp, + rmii_rxd0_mfp, rmii_ref_clk_mfp, i2s_d0_mfp, i2s_pcm1_mfp, + i2s0_pcm0_mfp, i2s1_pcm0_mfp, i2s_d1_mfp, ks_in2_mfp, + ks_in1_mfp, ks_in0_mfp, ks_in3_mfp, ks_out0_mfp, + ks_out1_mfp, ks_out2_mfp, lvds_o_pn_mfp, dsi_dn0_mfp, + dsi_dp2_mfp, lcd0_d17_mfp, dsi_dp3_mfp, dsi_dn3_mfp, + dsi_dp0_mfp, lvds_ee_pn_mfp, spi0_i2c_pcm_mfp, + spi0_i2s_pcm_mfp, dsi_dnp1_cp_mfp, lvds_e_pn_mfp, + dsi_dn2_mfp, uart2_rtsb_mfp, uart2_ctsb_mfp, uart3_rtsb_mfp, + uart3_ctsb_mfp, sd0_d0_mfp, sd0_d1_mfp, sd0_d2_d3_mfp, + sd1_d0_d3_mfp, sd0_cmd_mfp, sd0_clk_mfp, sd1_cmd_mfp, + uart0_rx_mfp, clko_25m_mfp, csi_cn_cp_mfp, sens0_ckout_mfp, + uart0_tx_mfp, i2c0_mfp, csi_dn_dp_mfp, sen0_pclk_mfp, + pcm1_in_mfp, pcm1_clk_mfp, pcm1_sync_mfp, pcm1_out_mfp, + dnand_data_wr_mfp, dnand_acle_ce0_mfp, nand_ceb2_mfp, + nand_ceb3_mfp] minItems: 1 maxItems: 32 @@ -100,10 +100,10 @@ patternProperties: Specify the alternative function to be configured for the given gpio pin groups. enum: [nor, eth_rmii, eth_smii, spi0, spi1, spi2, spi3, sens0, - sens1, uart0, uart1, uart2, uart3, uart4, uart5, uart6, i2s0, - i2s1, pcm1, pcm0, ks, jtag, pwm0, pwm1, pwm2, pwm3, pwm4, pwm5, - p0, sd0, sd1, sd2, i2c0, i2c1, i2c3, dsi, lvds, usb30, clko_25m, - mipi_csi, nand, spdif, ts, lcd0] + sens1, uart0, uart1, uart2, uart3, uart4, uart5, uart6, i2s0, + i2s1, pcm1, pcm0, ks, jtag, pwm0, pwm1, pwm2, pwm3, pwm4, pwm5, + p0, sd0, sd1, sd2, i2c0, i2c1, i2c3, dsi, lvds, usb30, clko_25m, + mipi_csi, nand, spdif, ts, lcd0] required: - groups @@ -126,14 +126,14 @@ patternProperties: items: oneOf: - enum: [sirq_drv, rmii_txd01_txen_drv, rmii_rxer_drv, - rmii_crs_drv, rmii_rxd10_drv, rmii_ref_clk_drv, - smi_mdc_mdio_drv, i2s_d0_drv, i2s_bclk0_drv, i2s3_drv, - i2s13_drv, pcm1_drv, ks_in_drv, ks_out_drv, lvds_all_drv, - lcd_dsi_drv, dsi_drv, sd0_d0_d3_drv, sd1_d0_d3_drv, - sd0_cmd_drv, sd0_clk_drv, sd1_cmd_drv, sd1_clk_drv, - spi0_all_drv, uart0_rx_drv, uart0_tx_drv, uart2_all_drv, - i2c0_all_drv, i2c12_all_drv, sens0_pclk_drv, - sens0_ckout_drv, uart3_all_drv] + rmii_crs_drv, rmii_rxd10_drv, rmii_ref_clk_drv, + smi_mdc_mdio_drv, i2s_d0_drv, i2s_bclk0_drv, i2s3_drv, + i2s13_drv, pcm1_drv, ks_in_drv, ks_out_drv, lvds_all_drv, + lcd_dsi_drv, dsi_drv, sd0_d0_d3_drv, sd1_d0_d3_drv, + sd0_cmd_drv, sd0_clk_drv, sd1_cmd_drv, sd1_clk_drv, + spi0_all_drv, uart0_rx_drv, uart0_tx_drv, uart2_all_drv, + i2c0_all_drv, i2c12_all_drv, sens0_pclk_drv, + sens0_ckout_drv, uart3_all_drv] minItems: 1 maxItems: 32 @@ -144,29 +144,29 @@ patternProperties: items: oneOf: - enum: [dnand_dqs, dnand_dqsn, eth_txd0, eth_txd1, eth_txen, - eth_rxer, eth_crs_dv, eth_rxd1, eth_rxd0, eth_ref_clk, - eth_mdc, eth_mdio, sirq0, sirq1, sirq2, i2s_d0, i2s_bclk0, - i2s_lrclk0, i2s_mclk0, i2s_d1, i2s_bclk1, i2s_lrclk1, - i2s_mclk1, ks_in0, ks_in1, ks_in2, ks_in3, ks_out0, ks_out1, - ks_out2, lvds_oep, lvds_oen, lvds_odp, lvds_odn, lvds_ocp, - lvds_ocn, lvds_obp, lvds_obn, lvds_oap, lvds_oan, lvds_eep, - lvds_een, lvds_edp, lvds_edn, lvds_ecp, lvds_ecn, lvds_ebp, - lvds_ebn, lvds_eap, lvds_ean, lcd0_d18, lcd0_d17, dsi_dp3, - dsi_dn3, dsi_dp1, dsi_dn1, dsi_cp, dsi_cn, dsi_dp0, dsi_dn0, - dsi_dp2, dsi_dn2, sd0_d0, sd0_d1, sd0_d2, sd0_d3, sd1_d0, - sd1_d1, sd1_d2, sd1_d3, sd0_cmd, sd0_clk, sd1_cmd, sd1_clk, - spi0_sclk, spi0_ss, spi0_miso, spi0_mosi, uart0_rx, - uart0_tx, i2c0_sclk, i2c0_sdata, sensor0_pclk, - sensor0_ckout, dnand_ale, dnand_cle, dnand_ceb0, dnand_ceb1, - dnand_ceb2, dnand_ceb3, uart2_rx, uart2_tx, uart2_rtsb, - uart2_ctsb, uart3_rx, uart3_tx, uart3_rtsb, uart3_ctsb, - pcm1_in, pcm1_clk, pcm1_sync, pcm1_out, i2c1_sclk, - i2c1_sdata, i2c2_sclk, i2c2_sdata, csi_dn0, csi_dp0, - csi_dn1, csi_dp1, csi_dn2, csi_dp2, csi_dn3, csi_dp3, - csi_cn, csi_cp, dnand_d0, dnand_d1, dnand_d2, dnand_d3, - dnand_d4, dnand_d5, dnand_d6, dnand_d7, dnand_rb, dnand_rdb, - dnand_rdbn, dnand_wrb, porb, clko_25m, bsel, pkg0, pkg1, - pkg2, pkg3] + eth_rxer, eth_crs_dv, eth_rxd1, eth_rxd0, eth_ref_clk, + eth_mdc, eth_mdio, sirq0, sirq1, sirq2, i2s_d0, i2s_bclk0, + i2s_lrclk0, i2s_mclk0, i2s_d1, i2s_bclk1, i2s_lrclk1, + i2s_mclk1, ks_in0, ks_in1, ks_in2, ks_in3, ks_out0, ks_out1, + ks_out2, lvds_oep, lvds_oen, lvds_odp, lvds_odn, lvds_ocp, + lvds_ocn, lvds_obp, lvds_obn, lvds_oap, lvds_oan, lvds_eep, + lvds_een, lvds_edp, lvds_edn, lvds_ecp, lvds_ecn, lvds_ebp, + lvds_ebn, lvds_eap, lvds_ean, lcd0_d18, lcd0_d17, dsi_dp3, + dsi_dn3, dsi_dp1, dsi_dn1, dsi_cp, dsi_cn, dsi_dp0, dsi_dn0, + dsi_dp2, dsi_dn2, sd0_d0, sd0_d1, sd0_d2, sd0_d3, sd1_d0, + sd1_d1, sd1_d2, sd1_d3, sd0_cmd, sd0_clk, sd1_cmd, sd1_clk, + spi0_sclk, spi0_ss, spi0_miso, spi0_mosi, uart0_rx, + uart0_tx, i2c0_sclk, i2c0_sdata, sensor0_pclk, + sensor0_ckout, dnand_ale, dnand_cle, dnand_ceb0, dnand_ceb1, + dnand_ceb2, dnand_ceb3, uart2_rx, uart2_tx, uart2_rtsb, + uart2_ctsb, uart3_rx, uart3_tx, uart3_rtsb, uart3_ctsb, + pcm1_in, pcm1_clk, pcm1_sync, pcm1_out, i2c1_sclk, + i2c1_sdata, i2c2_sclk, i2c2_sdata, csi_dn0, csi_dp0, + csi_dn1, csi_dp1, csi_dn2, csi_dp2, csi_dn3, csi_dp3, + csi_cn, csi_cp, dnand_d0, dnand_d1, dnand_d2, dnand_d3, + dnand_d4, dnand_d5, dnand_d6, dnand_d7, dnand_rb, dnand_rdb, + dnand_rdbn, dnand_wrb, porb, clko_25m, bsel, pkg0, pkg1, + pkg2, pkg3] minItems: 1 maxItems: 64 diff --git a/Documentation/devicetree/bindings/pinctrl/pinctrl-mt8192.yaml b/Documentation/devicetree/bindings/pinctrl/pinctrl-mt8192.yaml index 5556def6b99b..c4c071211611 100644 --- a/Documentation/devicetree/bindings/pinctrl/pinctrl-mt8192.yaml +++ b/Documentation/devicetree/bindings/pinctrl/pinctrl-mt8192.yaml @@ -106,7 +106,7 @@ patternProperties: required: - pinmux - additionalProperties: false + additionalProperties: false required: - compatible diff --git a/Documentation/devicetree/bindings/pinctrl/qcom,msm8226-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/qcom,msm8226-pinctrl.yaml index 1f0f5757f9e1..040d2ada3669 100644 --- a/Documentation/devicetree/bindings/pinctrl/qcom,msm8226-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/qcom,msm8226-pinctrl.yaml @@ -71,9 +71,9 @@ patternProperties: Specify the alternative function to be configured for the specified pins. Functions are only valid for gpio pins. enum: [ gpio, cci_i2c0, blsp_uim1, blsp_uim2, blsp_uim3, blsp_uim5, - blsp_i2c1, blsp_i2c2, blsp_i2c3, blsp_i2c5, blsp_spi1, - blsp_spi2, blsp_spi3, blsp_spi5, blsp_uart1, blsp_uart2, - blsp_uart3, blsp_uart5, cam_mclk0, cam_mclk1, wlan ] + blsp_i2c1, blsp_i2c2, blsp_i2c3, blsp_i2c5, blsp_spi1, + blsp_spi2, blsp_spi3, blsp_spi5, blsp_uart1, blsp_uart2, + blsp_uart3, blsp_uart5, cam_mclk0, cam_mclk1, wlan ] drive-strength: enum: [2, 4, 6, 8, 10, 12, 14, 16] diff --git a/Documentation/devicetree/bindings/pinctrl/toshiba,visconti-pinctrl.yaml b/Documentation/devicetree/bindings/pinctrl/toshiba,visconti-pinctrl.yaml index d0d1a01140ea..9f1dab0c2430 100644 --- a/Documentation/devicetree/bindings/pinctrl/toshiba,visconti-pinctrl.yaml +++ b/Documentation/devicetree/bindings/pinctrl/toshiba,visconti-pinctrl.yaml @@ -40,24 +40,24 @@ patternProperties: Function to mux. $ref: "/schemas/types.yaml#/definitions/string" enum: [i2c0, i2c1, i2c2, i2c3, i2c4, i2c5, i2c6, i2c7, i2c8, - spi0, spi1, spi2, spi3, spi4, spi5, spi6, - uart0, uart1, uart2, uart3, pwm, pcmif_out, pcmif_in] + spi0, spi1, spi2, spi3, spi4, spi5, spi6, + uart0, uart1, uart2, uart3, pwm, pcmif_out, pcmif_in] groups: description: Name of the pin group to use for the functions. $ref: "/schemas/types.yaml#/definitions/string" enum: [i2c0_grp, i2c1_grp, i2c2_grp, i2c3_grp, i2c4_grp, - i2c5_grp, i2c6_grp, i2c7_grp, i2c8_grp, - spi0_grp, spi0_cs0_grp, spi0_cs1_grp, spi0_cs2_grp, - spi1_grp, spi2_grp, spi3_grp, spi4_grp, spi5_grp, spi6_grp, - uart0_grp, uart1_grp, uart2_grp, uart3_grp, - pwm0_gpio4_grp, pwm0_gpio8_grp, pwm0_gpio12_grp, - pwm0_gpio16_grp, pwm1_gpio5_grp, pwm1_gpio9_grp, - pwm1_gpio13_grp, pwm1_gpio17_grp, pwm2_gpio6_grp, - pwm2_gpio10_grp, pwm2_gpio14_grp, pwm2_gpio18_grp, - pwm3_gpio7_grp, pwm3_gpio11_grp, pwm3_gpio15_grp, - pwm3_gpio19_grp, pcmif_out_grp, pcmif_in_grp] + i2c5_grp, i2c6_grp, i2c7_grp, i2c8_grp, + spi0_grp, spi0_cs0_grp, spi0_cs1_grp, spi0_cs2_grp, + spi1_grp, spi2_grp, spi3_grp, spi4_grp, spi5_grp, spi6_grp, + uart0_grp, uart1_grp, uart2_grp, uart3_grp, + pwm0_gpio4_grp, pwm0_gpio8_grp, pwm0_gpio12_grp, + pwm0_gpio16_grp, pwm1_gpio5_grp, pwm1_gpio9_grp, + pwm1_gpio13_grp, pwm1_gpio17_grp, pwm2_gpio6_grp, + pwm2_gpio10_grp, pwm2_gpio14_grp, pwm2_gpio18_grp, + pwm3_gpio7_grp, pwm3_gpio11_grp, pwm3_gpio15_grp, + pwm3_gpio19_grp, pcmif_out_grp, pcmif_in_grp] drive-strength: enum: [2, 4, 6, 8, 16, 24, 32] diff --git a/Documentation/devicetree/bindings/power/reset/reboot-mode.yaml b/Documentation/devicetree/bindings/power/reset/reboot-mode.yaml index a6c91026d4cc..a2dca38577f7 100644 --- a/Documentation/devicetree/bindings/power/reset/reboot-mode.yaml +++ b/Documentation/devicetree/bindings/power/reset/reboot-mode.yaml @@ -28,9 +28,9 @@ description: | properties: mode-normal: - $ref: /schemas/types.yaml#/definitions/uint32 - description: | - Default value to set on a reboot if no command was provided. + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Default value to set on a reboot if no command was provided. patternProperties: "^mode-.*$": diff --git a/Documentation/devicetree/bindings/power/supply/ingenic,battery.yaml b/Documentation/devicetree/bindings/power/supply/ingenic,battery.yaml index 867e3e6b7e80..76c227a7cd5c 100644 --- a/Documentation/devicetree/bindings/power/supply/ingenic,battery.yaml +++ b/Documentation/devicetree/bindings/power/supply/ingenic,battery.yaml @@ -15,10 +15,10 @@ properties: oneOf: - const: ingenic,jz4740-battery - items: - - enum: - - ingenic,jz4725b-battery - - ingenic,jz4770-battery - - const: ingenic,jz4740-battery + - enum: + - ingenic,jz4725b-battery + - ingenic,jz4770-battery + - const: ingenic,jz4740-battery io-channels: maxItems: 1 diff --git a/Documentation/devicetree/bindings/power/supply/summit,smb347-charger.yaml b/Documentation/devicetree/bindings/power/supply/summit,smb347-charger.yaml index 193a23af2007..983fc215c1e5 100644 --- a/Documentation/devicetree/bindings/power/supply/summit,smb347-charger.yaml +++ b/Documentation/devicetree/bindings/power/supply/summit,smb347-charger.yaml @@ -84,12 +84,12 @@ allOf: then: properties: summit,mains-current-limit-microamp: - enum: [ 300000, 500000, 700000, 1000000, - 1500000, 1800000, 2000000] + enum: [ 300000, 500000, 700000, 1000000, + 1500000, 1800000, 2000000] summit,usb-current-limit-microamp: - enum: [ 300000, 500000, 700000, 1000000, - 1500000, 1800000, 2000000] + enum: [ 300000, 500000, 700000, 1000000, + 1500000, 1800000, 2000000] summit,charge-current-compensation-microamp: enum: [200000, 450000, 600000, 900000] @@ -97,12 +97,12 @@ allOf: else: properties: summit,mains-current-limit-microamp: - enum: [ 300000, 500000, 700000, 900000, 1200000, - 1500000, 1800000, 2000000, 2200000, 2500000] + enum: [ 300000, 500000, 700000, 900000, 1200000, + 1500000, 1800000, 2000000, 2200000, 2500000] summit,usb-current-limit-microamp: - enum: [ 300000, 500000, 700000, 900000, 1200000, - 1500000, 1800000, 2000000, 2200000, 2500000] + enum: [ 300000, 500000, 700000, 900000, 1200000, + 1500000, 1800000, 2000000, 2200000, 2500000] summit,charge-current-compensation-microamp: enum: [250000, 700000, 900000, 1200000] diff --git a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml index 3f4a1939554d..efc0198eeb74 100644 --- a/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml +++ b/Documentation/devicetree/bindings/riscv/sifive-l2-cache.yaml @@ -25,8 +25,8 @@ select: properties: compatible: items: - - enum: - - sifive,fu540-c000-ccache + - enum: + - sifive,fu540-c000-ccache required: - compatible diff --git a/Documentation/devicetree/bindings/rng/imx-rng.yaml b/Documentation/devicetree/bindings/rng/imx-rng.yaml index 4ad1e456a801..07f6ff89bcc1 100644 --- a/Documentation/devicetree/bindings/rng/imx-rng.yaml +++ b/Documentation/devicetree/bindings/rng/imx-rng.yaml @@ -19,9 +19,9 @@ properties: - const: fsl,imx21-rnga - items: - enum: - - fsl,imx6sl-rngb - - fsl,imx6sll-rngb - - fsl,imx6ull-rngb + - fsl,imx6sl-rngb + - fsl,imx6sll-rngb + - fsl,imx6ull-rngb - const: fsl,imx25-rngb - const: fsl,imx35-rngc diff --git a/Documentation/devicetree/bindings/serial/fsl-imx-uart.yaml b/Documentation/devicetree/bindings/serial/fsl-imx-uart.yaml index 9ff85bc6859c..9702c07a6b6c 100644 --- a/Documentation/devicetree/bindings/serial/fsl-imx-uart.yaml +++ b/Documentation/devicetree/bindings/serial/fsl-imx-uart.yaml @@ -20,30 +20,30 @@ properties: - const: fsl,imx21-uart - items: - enum: - - fsl,imx25-uart - - fsl,imx27-uart - - fsl,imx31-uart - - fsl,imx35-uart - - fsl,imx50-uart - - fsl,imx51-uart - - fsl,imx53-uart - - fsl,imx6q-uart + - fsl,imx25-uart + - fsl,imx27-uart + - fsl,imx31-uart + - fsl,imx35-uart + - fsl,imx50-uart + - fsl,imx51-uart + - fsl,imx53-uart + - fsl,imx6q-uart - const: fsl,imx21-uart - items: - enum: - - fsl,imx6sl-uart - - fsl,imx6sll-uart - - fsl,imx6sx-uart + - fsl,imx6sl-uart + - fsl,imx6sll-uart + - fsl,imx6sx-uart - const: fsl,imx6q-uart - const: fsl,imx21-uart - items: - enum: - - fsl,imx6ul-uart - - fsl,imx7d-uart - - fsl,imx8mm-uart - - fsl,imx8mn-uart - - fsl,imx8mp-uart - - fsl,imx8mq-uart + - fsl,imx6ul-uart + - fsl,imx7d-uart + - fsl,imx8mm-uart + - fsl,imx8mn-uart + - fsl,imx8mp-uart + - fsl,imx8mq-uart - const: fsl,imx6q-uart reg: diff --git a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml index 7d8bd4e14434..4a2129005c0f 100644 --- a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml @@ -10,8 +10,8 @@ maintainers: - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> description: - The Microchip Sony/Philips Digital Interface Receiver is a - serial port compliant with the IEC-60958 standard. + The Microchip Sony/Philips Digital Interface Receiver is a serial port + compliant with the IEC-60958 standard. properties: "#sound-dai-cells": diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml index a03b0b871fc9..bdfb63387c53 100644 --- a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -10,8 +10,8 @@ maintainers: - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> description: - The Microchip Sony/Philips Digital Interface Transmitter is a - serial port compliant with the IEC-60958 standard. + The Microchip Sony/Philips Digital Interface Transmitter is a serial port + compliant with the IEC-60958 standard. properties: "#sound-dai-cells": diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index f6f9fb49f385..1e23c0e20bc1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -26,8 +26,10 @@ properties: reg: maxItems: 2 description: LPAIF core registers + reg-names: - maxItems: 2 + maxItems: 2 + clocks: minItems: 3 maxItems: 6 @@ -39,8 +41,10 @@ properties: interrupts: maxItems: 2 description: LPAIF DMA buffer interrupt + interrupt-names: maxItems: 2 + qcom,adsp: $ref: /schemas/types.yaml#/definitions/phandle description: Phandle for the audio DSP node @@ -141,31 +145,31 @@ allOf: properties: clock-names: oneOf: - - items: #for I2S - - const: pcnoc-sway-clk - - const: audio-core - - const: mclk0 - - const: pcnoc-mport-clk - - const: mi2s-bit-clk0 - - const: mi2s-bit-clk1 - - items: #for HDMI - - const: pcnoc-sway-clk - - const: audio-core - - const: pcnoc-mport-clk + - items: #for I2S + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + - items: #for HDMI + - const: pcnoc-sway-clk + - const: audio-core + - const: pcnoc-mport-clk reg-names: anyOf: - items: #for I2S - - const: lpass-lpaif + - const: lpass-lpaif - items: #for I2S and HDMI - - const: lpass-hdmiif - - const: lpass-lpaif + - const: lpass-hdmiif + - const: lpass-lpaif interrupt-names: anyOf: - items: #for I2S - - const: lpass-irq-lpaif + - const: lpass-irq-lpaif - items: #for I2S and HDMI - - const: lpass-irq-lpaif - - const: lpass-irq-hdmi + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi required: - iommus - power-domains diff --git a/Documentation/devicetree/bindings/timer/arm,sp804.yaml b/Documentation/devicetree/bindings/timer/arm,sp804.yaml index e35d3053250a..960e2bd66a97 100644 --- a/Documentation/devicetree/bindings/timer/arm,sp804.yaml +++ b/Documentation/devicetree/bindings/timer/arm,sp804.yaml @@ -33,8 +33,8 @@ properties: compatible: items: - enum: - - arm,sp804 - - hisilicon,sp804 + - arm,sp804 + - hisilicon,sp804 - const: arm,primecell interrupts: @@ -58,11 +58,11 @@ properties: clock is used for all clock inputs. oneOf: - items: - - description: clock for timer 1 - - description: clock for timer 2 - - description: bus clock + - description: clock for timer 1 + - description: clock for timer 2 + - description: bus clock - items: - - description: unified clock for both timers and the bus + - description: unified clock for both timers and the bus clock-names: true # The original binding did not specify any clock names, and there is no diff --git a/Documentation/devicetree/bindings/usb/cdns,usb3.yaml b/Documentation/devicetree/bindings/usb/cdns,usb3.yaml index ac20b98e9910..d6af2794d444 100644 --- a/Documentation/devicetree/bindings/usb/cdns,usb3.yaml +++ b/Documentation/devicetree/bindings/usb/cdns,usb3.yaml @@ -44,8 +44,8 @@ properties: enum: [super-speed, high-speed, full-speed] phys: - minItems: 1 - maxItems: 2 + minItems: 1 + maxItems: 2 phy-names: minItems: 1 diff --git a/Documentation/devicetree/bindings/usb/ti,hd3ss3220.yaml b/Documentation/devicetree/bindings/usb/ti,hd3ss3220.yaml index 5fe9e6211ba2..52ceb07294a3 100644 --- a/Documentation/devicetree/bindings/usb/ti,hd3ss3220.yaml +++ b/Documentation/devicetree/bindings/usb/ti,hd3ss3220.yaml @@ -17,7 +17,7 @@ description: |- properties: compatible: - const: ti,hd3ss3220 + const: ti,hd3ss3220 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/w1/fsl-imx-owire.yaml b/Documentation/devicetree/bindings/w1/fsl-imx-owire.yaml index 1aaf3e768c81..55adea827c34 100644 --- a/Documentation/devicetree/bindings/w1/fsl-imx-owire.yaml +++ b/Documentation/devicetree/bindings/w1/fsl-imx-owire.yaml @@ -15,10 +15,10 @@ properties: - const: fsl,imx21-owire - items: - enum: - - fsl,imx27-owire - - fsl,imx50-owire - - fsl,imx51-owire - - fsl,imx53-owire + - fsl,imx27-owire + - fsl,imx50-owire + - fsl,imx51-owire + - fsl,imx53-owire - const: fsl,imx21-owire reg: -- 2.25.1

6 5

[PATCH v2 0/4] HDA controller and PM cleanups
by Kai-Heng Feng 27 Oct '20

27 Oct '20

While working on the issue "ALSA: hda: fix jack detection with Realtek codecs when in D3", I've found using pm_runtime_force_{suspend,resume}() have surprising behavior, specifically, pm_runtime_need_not_resume() uses pm_runtime_need_not_resume() to avoid calling resume callback, so jackpoll was used to really power up the codec. We can use direct-complete to do the keep the codec suspended throughout the system PM flow, namely, keep the codec suspended all the way, unless the codec needs to be woken up after resume. For HDA controller, PCI core may enable direct-complete for it if conditions are met. So make runtime and system PM distinctive to always apply correct wake up setting. At least point, hopefully all runtime PM issues are solved, let's enable runtime PM by default again. Kai-Heng Feng (4): ALSA: hda: Refactor codec PM to use direct-complete optimization ALSA: hda: Stop mangling PCI IRQ ALSA: hda: Separate runtime and system suspend ALSA: hda: Reinstate runtime_allow() for all hda controllers sound/pci/hda/hda_codec.c | 45 +++++++++++++------- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 77 ++++++++++++++++------------------ 3 files changed, 67 insertions(+), 58 deletions(-) -- 2.17.1

3 12

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2
by Keith Winstein 27 Oct '20

27 Oct '20

Hi folks, We're using a Zoom UAC-2 USB audio interface, which provides an async playback endpoint ("1 OUT (ASYNC)") and an async capture endpoint ("2 IN (ASYNC)"), both with 2-channel S32_LE in 44100, 48000, 88200, 96000, 176400, or 192000 samples per second. For whatever reason, the explicit feedback on the playback endpoint seems a bit unstable and also biased a little high. alsaloop -t 1000 gets repeated playback underruns and starts trying to resample the audio to adjust the rate (even though it's just looping from capture back to playback on the same device!). I've found that using implicit feedback produces much more stable results and doesn't seem to hurt anything else (patch below). I wasn't sure if the other "implicit feedback quirk" devices in sound/usb/pcm.c are also ASYNC devices that happen to perform better using implicit feedback instead (in which case adding the Zoom UAC-2 to the list seems pretty routine), or if the others are SYNC devices that only need help identifying the right endpoint and ifnum -- i.e., how unorthodox is it to use implicit feedback on an ASYNC device? Hopefully not very. Please let me know if you'd like any additional information or testing. Best regards, Keith >From 2c8d691d39ab2824c344ac2562a1e8605b3384ba Mon Sep 17 00:00:00 2001 From: Keith Winstein <keithw(a)cs.stanford.edu> Date: Sun, 25 Oct 2020 22:05:47 -0700 Subject: [PATCH] ALSA: usb-audio: Add implicit feedback quirk for Zoom UAC-2 Signed-off-by: Keith Winstein <keithw(a)cs.stanford.edu> Tested-by: Keith Winstein <keithw(a)cs.stanford.edu> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..5723e8e69cf1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -352,6 +352,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1686, 0xf029): /* ZOOM UAC-2 */ + ep = 0x82; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; -- 2.25.1

2 4

RE: [PATCH] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 27 Oct '20

27 Oct '20

On 9/29/20 04:13 PM, Takashi Iwai wrote: >On Tue, 29 Sep 2020 03:51:35 +0200, >Gyeongtaek Lee wrote: >> >> On 9/28/20 11:35 PM, Pierre-Louis Bossart wrote: >> >On 9/28/20 6:13 AM, Jaroslav Kysela wrote: >> >> Dne 28. 09. 20 v 12:50 Gyeongtaek Lee napsal(a): >> >>> With a stream with low bitrate, user can't pause or resume the stream >> >>> near the end of the stream because current ALSA doesn't allow it. >> >>> If the stream has very low bitrate enough to store whole stream into >> >>> the buffer, user can't do anything except stop the stream and then >> >>> restart it from the first. >> >>> If pause and resume is allowed during draining, user experience can be >> >>> enhanced. >> >> >> >> It seems that we need a new state to handle the pause + drain condition for >> >> this case. >> >> >> >> With this proposed change, the pause state in drain is invisible. >> > >> >Indeed it's be much nicer to have a new state, e..g >> >SNDRV_PCM_STATE_DRAINING_PAUSED. >> Ok. I will add the new state. >> > >> >One concern is that states are defined in uapi/sound/asoc.h, so wouldn't >> >this have impacts on userspace as well? We'd change the value of >> >SNDRV_PCM_STATE_LAST. >> > >> I also agree that adding new state and increase LAST value in the header of uapi >> could be dangerous. So, I added it to comress_offload.h for now. >> It could be merged into snd_pcm_state_t in someday with big changes. >> Could you review the fixed patch below? > >Hrm, this resulted in rather more complex changes than the original >patch. It shows that introducing yet another state is no good idea >for this particular case. > >Since the possible application's behavior after this pause is as same >as the normal pause (i.e. either resuming pause or dropping), I find >it OK to take the original approach. Thank you for the review. I think that I should resend the original patch. Am I right? > > >thanks, > >Takashi > >> With a stream with low bitrate, user can't pause or resume the stream >> near the end of the stream because current ALSA doesn't allow it. >> If the stream has very low bitrate enough to store whole stream into >> the buffer, user can't do anything except stop the stream and then >> restart it from first. >> If pause, resume are allowed during draining, user experience can be >> enhanced. >> >> New state for pause during draining is defined in compress_offload.h for >> now. If PCM_STATEs in uapi/sound/asound.h is changed, pcm libraries and >> userspace application will be affected. >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org >> --- >> include/uapi/sound/compress_offload.h | 3 ++ >> sound/core/compress_offload.c | 47 ++++++++++++++++++++++----- >> 2 files changed, 41 insertions(+), 9 deletions(-) >> >> diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h >> index 7184265c0b0d..f30b9851d1d7 100644 >> --- a/include/uapi/sound/compress_offload.h >> +++ b/include/uapi/sound/compress_offload.h >> @@ -189,4 +189,7 @@ struct snd_compr_metadata { >> #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ >> #define SND_COMPR_TRIGGER_NEXT_TRACK 8 >> #define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 >> + >> +/* FIXME move this to asound.h */ >> +#define SNDRV_PCM_STATE_DRAINING_PAUSED (SNDRV_PCM_STATE_LAST + 1) >> #endif >> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c >> index 0e53f6f31916..58fbe0d99431 100644 >> --- a/sound/core/compress_offload.c >> +++ b/sound/core/compress_offload.c >> @@ -151,6 +151,7 @@ static int snd_compr_free(struct inode *inode, struct file *f) >> case SNDRV_PCM_STATE_RUNNING: >> case SNDRV_PCM_STATE_DRAINING: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); >> break; >> default: >> @@ -431,6 +432,7 @@ static __poll_t snd_compr_poll(struct file *f, poll_table *wait) >> case SNDRV_PCM_STATE_RUNNING: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> if (avail >= stream->runtime->fragment_size) >> retval = snd_compr_get_poll(stream); >> break; >> @@ -708,11 +710,23 @@ static int snd_compr_pause(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_RUNNING: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + break; >> + case SNDRV_PCM_STATE_DRAINING: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> + if (!retval) >> + stream->runtime->state = >> + SNDRV_PCM_STATE_DRAINING_PAUSED; >> + break; >> + default: >> return -EPERM; >> - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> - if (!retval) >> - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + } >> return retval; >> } >> >> @@ -720,11 +734,22 @@ static int snd_compr_resume(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_PAUSED: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; >> + break; >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; >> + break; >> + default: >> return -EPERM; >> - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> - if (!retval) >> - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; >> + } >> return retval; >> } >> >> @@ -835,7 +860,9 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) >> */ >> >> ret = wait_event_interruptible(stream->runtime->sleep, >> - (stream->runtime->state != SNDRV_PCM_STATE_DRAINING)); >> + (stream->runtime->state != SNDRV_PCM_STATE_DRAINING) && >> + (stream->runtime->state != >> + SNDRV_PCM_STATE_DRAINING_PAUSED)); >> if (ret == -ERESTARTSYS) >> pr_debug("wait aborted by a signal\n"); >> else if (ret) >> @@ -857,6 +884,7 @@ static int snd_compr_drain(struct snd_compr_stream *stream) >> case SNDRV_PCM_STATE_SETUP: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> return -EPERM; >> case SNDRV_PCM_STATE_XRUN: >> return -EPIPE; >> @@ -909,6 +937,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) >> case SNDRV_PCM_STATE_SETUP: >> case SNDRV_PCM_STATE_PREPARED: >> case SNDRV_PCM_STATE_PAUSED: >> + case SNDRV_PCM_STATE_DRAINING_PAUSED: >> return -EPERM; >> case SNDRV_PCM_STATE_XRUN: >> return -EPIPE; >> >> base-commit: a1b8638ba1320e6684aa98233c15255eb803fac7 >> -- >> 2.21.0 >> >> >

5 25

[PATCH AUTOSEL 5.8 035/132] ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
by Sasha Levin 27 Oct '20

27 Oct '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 6c63c954e1c52f1262f986f36d95f557c6f8fa94 ] When hda_codec_probe() doesn't initialize audio component, we disable the codec and keep going. However,the resources are not released. The child_count of SOF device is increased in snd_hdac_ext_bus_device_init but is not decrease in error case, so SOF can't get suspended. snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it gets a error. Now copy this behavior to release resources and decrease SOF device child_count to release SOF device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d7..c475955c6eeba 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -151,7 +151,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - return -ENOENT; + goto error; } hda_priv->need_display_power = true; } @@ -174,7 +174,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - ret = -ENOENT; + goto error; } return ret; -- 2.25.1

1 0

[PATCH AUTOSEL 5.9 043/147] ASoC: AMD: Clean kernel log from deferred probe error messages
by Sasha Levin 27 Oct '20

27 Oct '20

From: Akshu Agrawal <akshu.agrawal(a)amd.com> [ Upstream commit f7660445c8e7fda91e8b944128554249d886b1d4 ] While the driver waits for DAIs to be probed and retries probing, have the error messages at debug level instead of error. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Link: https://lore.kernel.org/r/20200826185454.5545-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/acp3x-rt5682-max9836.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af34..1a4e8ca0f99c2 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -472,12 +472,17 @@ static int acp3x_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); } - return 0; + + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.25.1

1 0

[PATCH] ASoC: amd: support other audio modes for raven
by Vijendar Mukunda 27 Oct '20

27 Oct '20

ACP supports different audio configurations other than I2S. This patch will fix acp driver probe failure for other audio configurations. Tested-by: Prike Liang <Prike.Liang(a)amd.com> Reviewed-by: Prike Liang <Prike.Liang(a)amd.com> Acked-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 31b797c..cd5b097 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -231,9 +231,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } break; default: - dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); - ret = -ENODEV; - goto disable_msi; + dev_info(&pci->dev, "ACP audio mode : %d\n", val); } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); -- 2.7.4

2 1

[PATCH] ASoC: jz4740-i2s: Remove manual DMA peripheral ID assignment
by Lars-Peter Clausen 27 Oct '20

27 Oct '20

All platforms that use the jz4740-i2s driver have been switched to devicetree for a while now and the assignment of the DMA peripheral ID is done in the devicetree. It is no longer necessary to manually assign the peripheral ID in the driver, so remove that. The DMA driver does not even look at the value assigned in the driver anymore and always uses the value provided by the devicetree. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..e8bc7ca5ee5e 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -26,9 +26,6 @@ #include "jz4740-i2s.h" -#define JZ4740_DMA_TYPE_AIC_TRANSMIT 24 -#define JZ4740_DMA_TYPE_AIC_RECEIVE 25 - #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 #define JZ_REG_AIC_I2S_FMT 0x10 @@ -377,13 +374,11 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_data = &i2s->playback_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ dma_data = &i2s->capture_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } -- 2.20.1

3 2

[PATCH] ASoC: wm5102: Use get_unaligned_be16() for dac_comp_coeff
by Richard Fitzgerald 27 Oct '20

27 Oct '20

Replace the two-step copy-and-convert in wm5102_out_comp_coeff_put() with get_unaligned_be16(). Apart from looking nicer, it avoids this sparse warning: wm5102.c:687:35: sparse: sparse: cast to restricted __be16 Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm5102.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 2ed3fa67027d..70d353b63fe0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -682,9 +682,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct arizona *arizona = dev_get_drvdata(component->dev->parent); mutex_lock(&arizona->dac_comp_lock); - memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, - sizeof(arizona->dac_comp_coeff)); - arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); mutex_unlock(&arizona->dac_comp_lock); return 0; base-commit: 2ebefde9de62a0828936e47b62975501993b5ee9 -- 2.20.1

2 1

[PATCH 1/2] ASoC: dt-bindings: fsl_spdif: Add new compatible string for i.MX8QM
by Shengjiu Wang 27 Oct '20

27 Oct '20

Add new compatible string "fsl,imx8qm-spdif" for supporting spdif module on i.MX8QM. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 2ac671f5cb9b..50449b6d1048 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx35-spdif - fsl,vf610-spdif - fsl,imx6sx-spdif + - fsl,imx8qm-spdif reg: maxItems: 1 -- 2.27.0

4 4

[PATCH 0/6] ASoC: merge soc_pcm_hw_param() rollback and soc_pcm_hw_free()
by Kuninori Morimoto 27 Oct '20

27 Oct '20

Hi Mark soc_pcm_hw_params() does rollback when failed (A), but, it is almost same as soc_pcm_hw_free(). static int soc_pcm_hw_params(xxx) { ... if (ret < 0) goto xxx_err; ... return ret; ^ component_err: | ... | interface_err: (A) ... | codec_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_hw_free() and rollback. Kuninori Morimoto (6): ASoC: soc.h: remove for_each_rtd_dais_rollback() ASoC: soc-pcm: move soc_pcm_hw_free() next to soc_pcm_hw_params() ASoC: soc-link: add mark for snd_soc_link_hw_params/free() ASoC: soc-component: add mark for snd_soc_pcm_component_hw_params/free() ASoC: soc-dai: add mark for snd_soc_dai_hw_params/free() ASoC: soc-pcm: add soc_pcm_hw_clean() and call it from soc_pcm_hw_params/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 7 +- sound/soc/soc-component.c | 19 ++--- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 12 +++- sound/soc/soc-pcm.c | 131 ++++++++++++++-------------------- 9 files changed, 97 insertions(+), 102 deletions(-) -- 2.25.1

2 7

[PATCH v2 00/17] ASoC: sun8i-codec: support for AIF2 and AIF3
by Samuel Holland 27 Oct '20

27 Oct '20

This series adds support the other two AIFs present in the sun8i codec, which can be used for codec2codec DAI links. This series first fills out the DAI driver, removing assumptions that were made for AIF1 (16 bits, 2 channels, certain clock inversions). Some new logic is required to handle 3 DAIs and the ADC/DAC sharing the same clock. Finally, it adds the new DAIs, and hooks them up with DAPM widgets and routes per the hardware topology. To minimize the number of patches in this series, related device tree patches (increasing #sound-dai-cells, adding new DAI links) will be sent separately. Changes from v1: - Patches 1-8 from v1 (DAPM changes) were merged - Prefixed AIF constants with "SUN8I_CODEC_" [1, 7, 10, 16, 17] - Renamed variables in sun8i_codec_set_fmt for clarity [3] - Update sysclk->sysclk_rate if later calls to hw_params change the sample rate (thanks Chen-Yu for reminding me of this) [11] - Select COMMON_CLK for clk_set_rate_exclusive [12] - Add comments and hopefully clarify the clock protection logic [12] - Make the error message more concise and put it on one line [12] - Drop the "reg" variable holding SUN8I_AIF_CLK_CTRL(dai->id) [15] - Rename "div_reg" to "clk_reg" and adjust comments for clarity [17] - Improve the AIF2/AIF3 rate mismatch error message [17] - Minor grammar/wording fixes in commit messages [2, 4, 7, 8, 16, 17] - Added Maxime's Acked-by: [1-2, 4-9, 11, 13-14, 16] Samuel Holland (17): ASoC: sun8i-codec: Prepare to extend the DAI driver ASoC: sun8i-codec: Program DAI format before clock inversion ASoC: sun8i-codec: Enable all supported clock inversions ASoC: sun8i-codec: Use the provided word size ASoC: sun8i-codec: Round up the LRCK divisor ASoC: sun8i-codec: Correct the BCLK divisor calculation ASoC: sun8i-codec: Support the TDM slot binding ASoC: sun8i-codec: Enforce symmetric DAI parameters ASoC: sun8i-codec: Enable all supported sample rates ASoC: sun8i-codec: Automatically set the system sample rate ASoC: sun8i-codec: Constrain to compatible sample rates ASoC: sun8i-codec: Protect the clock rate while streams are open ASoC: sun8i-codec: Require an exact BCLK divisor match ASoC: sun8i-codec: Enable all supported PCM formats ASoC: sun8i-codec: Generalize AIF clock control ASoC: sun8i-codec: Add the AIF2 DAI, widgets, and routes ASoC: sun8i-codec: Add the AIF3 DAI, widgets, and routes sound/soc/sunxi/Kconfig | 1 + sound/soc/sunxi/sun8i-codec.c | 834 ++++++++++++++++++++++++++++------ 2 files changed, 699 insertions(+), 136 deletions(-) -- 2.26.2

3 23

[PATCH] ASoC: adau1977: remove platform data and move micbias bindings include
by Alexandru Ardelean 27 Oct '20

27 Oct '20

The change removes the platform_data include/definition. It only contains some values for the MICBIAS. These are moved into 'dt-bindings/sound/adi,adau1977.h' so that they can be used inside device-trees. When moving then, they need to be converted to pre-compiler defines, so that the DT compiler can understand them. The driver then, also needs to include the new 'dt-bindings/sound/adi,adau1977.h' file. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- include/dt-bindings/sound/adi,adau1977.h | 15 ++++++++ include/linux/platform_data/adau1977.h | 44 ------------------------ sound/soc/codecs/adau1977.c | 9 ++--- 3 files changed, 18 insertions(+), 50 deletions(-) create mode 100644 include/dt-bindings/sound/adi,adau1977.h delete mode 100644 include/linux/platform_data/adau1977.h diff --git a/include/dt-bindings/sound/adi,adau1977.h b/include/dt-bindings/sound/adi,adau1977.h new file mode 100644 index 000000000000..8eebec6570f2 --- /dev/null +++ b/include/dt-bindings/sound/adi,adau1977.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +#ifndef __DT_BINDINGS_ADI_ADAU1977_H__ +#define __DT_BINDINGS_ADI_ADAU1977_H__ + +#define ADAU1977_MICBIAS_5V0 0x0 +#define ADAU1977_MICBIAS_5V5 0x1 +#define ADAU1977_MICBIAS_6V0 0x2 +#define ADAU1977_MICBIAS_6V5 0x3 +#define ADAU1977_MICBIAS_7V0 0x4 +#define ADAU1977_MICBIAS_7V5 0x5 +#define ADAU1977_MICBIAS_8V0 0x6 +#define ADAU1977_MICBIAS_8V5 0x7 +#define ADAU1977_MICBIAS_9V0 0x8 + +#endif /* __DT_BINDINGS_ADI_ADAU1977_H__ */ diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h deleted file mode 100644 index 86667235077a..000000000000 --- a/include/linux/platform_data/adau1977.h +++ /dev/null @@ -1,44 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * ADAU1977/ADAU1978/ADAU1979 driver - * - * Copyright 2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - */ - -#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__ -#define __LINUX_PLATFORM_DATA_ADAU1977_H__ - -/** - * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting - * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V - * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V - * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V - * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V - * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V - * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V - * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V - * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V - * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V - */ -enum adau1977_micbias { - ADAU1977_MICBIAS_5V0 = 0x0, - ADAU1977_MICBIAS_5V5 = 0x1, - ADAU1977_MICBIAS_6V0 = 0x2, - ADAU1977_MICBIAS_6V5 = 0x3, - ADAU1977_MICBIAS_7V0 = 0x4, - ADAU1977_MICBIAS_7V5 = 0x5, - ADAU1977_MICBIAS_8V0 = 0x6, - ADAU1977_MICBIAS_8V5 = 0x7, - ADAU1977_MICBIAS_9V0 = 0x8, -}; - -/** - * struct adau1977_platform_data - Platform configuration data for the ADAU1977 - * @micbias: Specifies the voltage for the MICBIAS pin - */ -struct adau1977_platform_data { - enum adau1977_micbias micbias; -}; - -#endif diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 0a36e523584c..8260f49caa24 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -12,7 +12,6 @@ #include <linux/i2c.h> #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_data/adau1977.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -24,6 +23,8 @@ #include <sound/soc.h> #include <sound/tlv.h> +#include <dt-bindings/sound/adi,adau1977.h> + #include "adau1977.h" #define ADAU1977_REG_POWER 0x00 @@ -881,13 +882,9 @@ static const struct snd_soc_component_driver adau1977_component_driver = { static int adau1977_setup_micbias(struct adau1977 *adau1977) { - struct adau1977_platform_data *pdata = adau1977->dev->platform_data; unsigned int micbias; - if (pdata) - micbias = pdata->micbias; - else if (device_property_read_u32(adau1977->dev, "adi,micbias", - &micbias)) + if (device_property_read_u32(adau1977->dev, "adi,micbias", &micbias)) micbias = ADAU1977_MICBIAS_8V5; if (micbias > ADAU1977_MICBIAS_9V0) { -- 2.17.1

3 3

[PATCH v4 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 27 Oct '20

27 Oct '20

From: Viorel Suman <viorel.suman(a)nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file Changes since v2: - used devm_reset_control_get_exclusive instead of of_reset_control_get - moved reset_control_assert into runtime_suspend Changes since v3: - removed "firmware-name" DTS property from both documentation and source code by porting it into SoC specific 'compatible' data structure. .../devicetree/bindings/sound/fsl,xcvr.yaml | 104 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1359 +++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1741 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.26.2

4 5

[PATCH 0/8] use semicolons rather than commas to separate statements
by Julia Lawall 27 Oct '20

27 Oct '20

These patches replace commas by semicolons. This was done using the Coccinelle semantic patch (http://coccinelle.lip6.fr/) shown below. This semantic patch ensures that commas inside for loop headers will not be transformed. It also doesn't touch macro definitions. Coccinelle ensures that braces are added as needed when a single-statement branch turns into a multi-statement one. This semantic patch has a few false positives, for variable delcarations such as: LIST_HEAD(x), *y; The semantic patch could be improved to avoid these, but for the moment they have been removed manually (2 occurrences). // <smpl> @initialize:ocaml@ @@ let infunction p = (* avoid macros *) (List.hd p).current_element <> "something_else" let combined p1 p2 = (List.hd p1).line_end = (List.hd p2).line || (((List.hd p1).line_end < (List.hd p2).line) && ((List.hd p1).col < (List.hd p2).col)) @bad@ statement S; declaration d; position p; @@ S@p d // special cases where newlines are needed (hope for no more than 5) @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @@ expression e1,e2; statement S; position p != bad.p; position p1; position p2 : script:ocaml(p1) { infunction p1 && combined p1 p2 }; @@ - e1@p1,@S@p e2@p2; + e1; e2; @r@ expression e1,e2; statement S; position p != bad.p; @@ e1 ,@S@p e2; @@ expression e1,e2; position p1; position p2 : script:ocaml(p1) { infunction p1 && not(combined p1 p2) }; statement S; position r.p; @@ e1@p1 -,@S@p +; e2@p2 ... when any // </smpl> --- sound/firewire/fireworks/fireworks_pcm.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/soc/codecs/madera.c | 4 ++-- sound/soc/codecs/wm8350.c | 3 ++- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/soc-dapm.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- 9 files changed, 11 insertions(+), 10 deletions(-)

7 16

[PATCH v2 0/2] dt-bindings: stm32: convert audio dfsdm to json-schema
by Olivier Moysan 27 Oct '20

27 Oct '20

Some audio properties documented in st,stm32-adfsdm.txt are already documented in Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml bindings. Move remaining properties from st,stm32-adfsdm.txt to st,stm32-dfsdm-adc.yaml, and remove st,stm32-adfsdm.txt. Changes in v2: - Complete st,stm32-dfsdm-adc.yaml rather than converting st,stm32-adfsdm.txt Olivier Moysan (2): dt-bindings: stm32: dfsdm: update audio properties dt-bindings: stm32: dfsdm: remove stm32-adfsdm.txt binding .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 7 ++- .../bindings/sound/st,stm32-adfsdm.txt | 63 ------------------- 2 files changed, 5 insertions(+), 65 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt -- 2.17.1

3 6

[PATCH v4] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 27 Oct '20

27 Oct '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" Changes in v3: - move clocks properties for st,stm32h7-sai compatible, to 'else' clause Changes in v4: - fix dtbs_check errors --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 200 ++++++++++++++++++ 2 files changed, 200 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index c42b91e525fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/st,stm32-dma.yaml - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..6ad48c7681c1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,200 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + maxItems: 3 + + clock-names: + maxItems: 3 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + + - else: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai2: sai@4400b000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400b000 0x400>; + reg = <0x4400b000 0x4>, <0x4400b3f0 0x10>; + clocks = <&rcc SAI2>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "x8k", "x11k"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>; + pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>; + status = "okay"; + + sai2a: audio-controller@4400b004 { + #sound-dai-cells = <0>; + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1c>; + dmas = <&dmamux1 89 0x400 0x01>; + dma-names = "tx"; + clocks = <&rcc SAI2_K>; + clock-names = "sai_ck"; + status = "okay"; + }; + }; + +... -- 2.17.1

3 2

[PATCH] ASoC: pcm5102a: Make codec selectable
by Codrin Ciubotariu 27 Oct '20

27 Oct '20

The TI PCM5102A codec driver can be used with the generic sound card drivers, so it should be selectable. For example, with the addition of #sound-dai-cells = <0> property in DT, it can be used with simple/graph card drivers. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 34c6dd04b85a..5791b7056af6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1003,7 +1003,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.25.1

2 1

[PATCH v2] ASoC: fsl_spdif: Add support for higher sample rates
by Shengjiu Wang 27 Oct '20

27 Oct '20

Add 88200Hz and 176400Hz sample rates support for TX. Add 88200Hz, 176400Hz, 192000Hz sample rates support for RX. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- changes in v2 - reorder the signed-off - add acked-by Nicolin sound/soc/fsl/fsl_spdif.c | 16 +++++++++++++--- sound/soc/fsl/fsl_spdif.h | 9 ++++++++- 2 files changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d14f4076ead..1c030142556a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -459,10 +459,18 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, rate = SPDIF_TXRATE_48000; csfs = IEC958_AES3_CON_FS_48000; break; + case 88200: + rate = SPDIF_TXRATE_88200; + csfs = IEC958_AES3_CON_FS_88200; + break; case 96000: rate = SPDIF_TXRATE_96000; csfs = IEC958_AES3_CON_FS_96000; break; + case 176400: + rate = SPDIF_TXRATE_176400; + csfs = IEC958_AES3_CON_FS_176400; + break; case 192000: rate = SPDIF_TXRATE_192000; csfs = IEC958_AES3_CON_FS_192000; @@ -857,7 +865,7 @@ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 16000; - uinfo->value.integer.max = 96000; + uinfo->value.integer.max = 192000; return 0; } @@ -1175,7 +1183,8 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + 192000, }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; u32 arate; @@ -1235,7 +1244,8 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; + static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + 192000, }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; u64 savesub = 100000, ret; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index e6c61e07bc1a..d5f1dfd58740 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -163,7 +163,9 @@ enum spdif_txrate { SPDIF_TXRATE_32000 = 0, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, + SPDIF_TXRATE_88200, SPDIF_TXRATE_96000, + SPDIF_TXRATE_176400, SPDIF_TXRATE_192000, }; #define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1) @@ -177,15 +179,20 @@ enum spdif_txrate { #define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ SNDRV_PCM_RATE_192000) #define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_64000 | \ - SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) #define FSL_SPDIF_FORMATS_PLAYBACK (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ -- 2.27.0

2 1

[PATCH] ASoC: Intel: kbl_rt5663_max98927: Fix kabylake_ssp_fixup function
by Tomasz Figa 27 Oct '20

27 Oct '20

This is a copy of commit 5c5f1baee85a ("ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function") applied to the kbl_rt5663_max98927 board file. Original explanation of the change: kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Fixes a boot crash on a HP Chromebook x2: [ 16.582225] BUG: kernel NULL pointer dereference, address: 0000000000000050 [ 16.582231] #PF: supervisor read access in kernel mode [ 16.582233] #PF: error_code(0x0000) - not-present page [ 16.582234] PGD 0 P4D 0 [ 16.582238] Oops: 0000 [#1] PREEMPT SMP PTI [ 16.582241] CPU: 0 PID: 1980 Comm: cras Tainted: G C 5.4.58 #1 [ 16.582243] Hardware name: HP Soraka/Soraka, BIOS Google_Soraka.10431.75.0 08/30/2018 [ 16.582247] RIP: 0010:kabylake_ssp_fixup+0x19/0xbb [snd_soc_kbl_rt5663_max98927] [ 16.582250] Code: c6 6f c5 80 c0 44 89 f2 31 c0 e8 3e c9 4c d6 eb de 0f 1f 44 00 00 55 48 89 e5 41 57 41 56 53 48 89 f3 48 8b 46 c8 48 8b 4e d0 <48> 8b 49 10 4c 8b 78 10 4c 8b 31 4c 89 f7 48 c7 c6 4b c2 80 c0 e8 [ 16.582252] RSP: 0000:ffffaf7e81e0b958 EFLAGS: 00010282 [ 16.582254] RAX: ffffffff96f13e0d RBX: ffffaf7e81e0ba00 RCX: 0000000000000040 [ 16.582256] RDX: ffffaf7e81e0ba00 RSI: ffffaf7e81e0ba00 RDI: ffffa3b208558028 [ 16.582258] RBP: ffffaf7e81e0b970 R08: ffffa3b203b54160 R09: ffffaf7e81e0ba00 [ 16.582259] R10: 0000000000000000 R11: ffffffffc080b345 R12: ffffa3b209fb6e00 [ 16.582261] R13: ffffa3b1b1a47838 R14: ffffa3b1e6197f28 R15: ffffaf7e81e0ba00 [ 16.582263] FS: 00007eb3f25aaf80(0000) GS:ffffa3b236a00000(0000) knlGS:0000000000000000 [ 16.582265] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 16.582267] CR2: 0000000000000050 CR3: 0000000246bc8006 CR4: 00000000003606f0 [ 16.582269] Call Trace: [ 16.582275] snd_soc_link_be_hw_params_fixup+0x21/0x68 [ 16.582278] snd_soc_dai_hw_params+0x25/0x94 [ 16.582282] soc_pcm_hw_params+0x2d8/0x583 [ 16.582288] dpcm_be_dai_hw_params+0x172/0x29e [ 16.582291] dpcm_fe_dai_hw_params+0x9f/0x12f [ 16.582295] snd_pcm_hw_params+0x137/0x41c [ 16.582298] snd_pcm_hw_params_user+0x3c/0x71 [ 16.582301] snd_pcm_common_ioctl+0x2c6/0x565 [ 16.582304] snd_pcm_ioctl+0x32/0x36 [ 16.582307] do_vfs_ioctl+0x506/0x783 [ 16.582311] ksys_ioctl+0x58/0x83 [ 16.582313] __x64_sys_ioctl+0x1a/0x1e [ 16.582316] do_syscall_64+0x54/0x7e [ 16.582319] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [ 16.582322] RIP: 0033:0x7eb3f1886157 [ 16.582324] Code: 8a 66 90 48 8b 05 11 dd 2b 00 64 c7 00 26 00 00 00 48 c7 c0 ff ff ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 b8 10 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d e1 dc 2b 00 f7 d8 64 89 01 48 [ 16.582326] RSP: 002b:00007ffff7559818 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [ 16.582329] RAX: ffffffffffffffda RBX: 00005acc9188b140 RCX: 00007eb3f1886157 [ 16.582330] RDX: 00007ffff7559940 RSI: 00000000c2604111 RDI: 000000000000001e [ 16.582332] RBP: 00007ffff7559840 R08: 0000000000000004 R09: 0000000000000000 [ 16.582333] R10: 0000000000000000 R11: 0000000000000246 R12: 000000000000bb80 [ 16.582335] R13: 00005acc91702e80 R14: 00007ffff7559940 R15: 00005acc91702e80 [ 16.582337] Modules linked in: rfcomm cmac algif_hash algif_skcipher af_alg uinput hid_google_hammer snd_soc_kbl_rt5663_max98927 snd_soc_hdac_hdmi snd_soc_dmic snd_soc_skl_ssp_clk snd_soc_skl snd_soc_sst_ipc snd_soc_sst_dsp snd_soc_hdac_hda snd_soc_acpi_intel_match snd_soc_acpi snd_hda_ext_core snd_intel_dspcfg snd_hda_codec snd_hwdep snd_hda_core ipu3_cio2 ipu3_imgu(C) videobuf2_v4l2 videobuf2_common videobuf2_dma_sg videobuf2_memops snd_soc_rt5663 snd_soc_max98927 snd_soc_rl6231 ov5670 ov13858 acpi_als v4l2_fwnode dw9714 fuse xt_MASQUERADE iio_trig_sysfs cros_ec_light_prox cros_ec_sensors cros_ec_sensors_core cros_ec_sensors_ring industrialio_triggered_buffer kfifo_buf industrialio cros_ec_sensorhub cdc_ether usbnet btusb btrtl btintel btbcm bluetooth ecdh_generic ecc lzo_rle lzo_compress iwlmvm zram iwl7000_mac80211 r8152 mii iwlwifi cfg80211 joydev [ 16.584243] gsmi: Log Shutdown Reason 0x03 [ 16.584246] CR2: 0000000000000050 [ 16.584248] ---[ end trace c8511d090c11edff ]--- Suggested-by: Łukasz Majczak <lmajczak(a)google.com> Fixes: 2e5894d73789e ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Tomasz Figa <tfiga(a)chromium.org> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 ++++++++++++++++---- 1 file changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3ea4602dfb3e..9a4b3d0973f6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.28.0.1011.ga647a8990f-goog

3 5

[PATCH] ALSA: make snd_kcontrol_new name a normal string
by Arnd Bergmann 26 Oct '20

26 Oct '20

From: Arnd Bergmann <arnd(a)arndb.de> When building with W=2, there are lots of warnings about the snd_kcontrol_new name field being an array of 'unsigned char' but initialized to a string: include/sound/soc.h:93:48: warning: pointer targets in initialization of 'const unsigned char *' from 'char *' differ in signedness [-Wpointer-sign] Make it a regular 'char *' to avoid flooding the build log with this. Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/control.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/control.h b/include/sound/control.h index e128cff10dfa..77d9fa10812d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -42,7 +42,7 @@ struct snd_kcontrol_new { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - const unsigned char *name; /* ASCII name of item */ + const char *name; /* ASCII name of item */ unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ -- 2.27.0

2 3

[PATCH v3] ASoC: Intel: boards: Add CML_RT1015 m/c driver
by Keith Tzneg 26 Oct '20

26 Oct '20

From: Keith Tzeng <keith.tzeng(a)quanta.corp-partner.google.com> Machine driver to enable RT5682 on SSP0, DMIC, HDMI and RT1015 AMP on SSP1: Enabled 4 CH TDM playback with 24 bit data. Signed-off-by: Keith Tzeng <keith.tzeng(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1015_rt5682.c | 570 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 594 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1015_rt5682.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index c10c378..f674c56 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -496,6 +496,21 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_CML_RT1015_RT5682_MACH + tristate "CML with RT1015 and RT5682 in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC + select SND_SOC_RT1015 + select SND_SOC_RT5682_I2C + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC machine driver for SOF platform with + RT1015 + RT5682 I2S codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK if SND_SOC_SOF_JASPERLAKE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d2..73131cc 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -20,6 +20,7 @@ snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o +snd-soc-cml_rt1015_rt5682-objs := cml_rt1015_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o @@ -60,6 +61,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH) += snd-soc-cml_rt1011_rt5682.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1015_RT5682_MACH) += cml_rt1015_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o diff --git a/sound/soc/intel/boards/cml_rt1015_rt5682.c b/sound/soc/intel/boards/cml_rt1015_rt5682.c new file mode 100644 index 0000000..9117f52 --- /dev/null +++ b/sound/soc/intel/boards/cml_rt1015_rt5682.c @@ -0,0 +1,570 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2019 Intel Corporation. + +/* + * Intel Cometlake I2S Machine driver for RT1015 + RT5682 codec + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/dmi.h> +#include <linux/slab.h> +#include <asm/cpu_device_id.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/rt5682.h> +#include <sound/soc-acpi.h> +#include "../../codecs/rt1015.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 24Mhz clock to codec as I2S MCLK */ +#define CML_PLAT_CLK 24000000 +#define CML_RT1015_CODEC_DAI "rt1015-aif" +#define CML_RT5682_CODEC_DAI "rt5682-aif1" +#define NAME_SIZE 32 + +#define SOF_RT1015_SPEAKER_WL BIT(0) +#define SOF_RT1015_SPEAKER_WR BIT(1) +#define SOF_RT1015_SPEAKER_TL BIT(2) +#define SOF_RT1015_SPEAKER_TR BIT(3) +#define SPK_CH 4 + +/* Default: Woofer speakers */ +static unsigned long sof_rt1015_quirk = SOF_RT1015_SPEAKER_WL | + SOF_RT1015_SPEAKER_WR; + +static int sof_rt1015_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt1015_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt1015_quirk_table[] = { + { + .callback = sof_rt1015_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Dooly"), + }, + .driver_data = (void *)(SOF_RT1015_SPEAKER_WL | SOF_RT1015_SPEAKER_WR), + }, + { + } +}; + +static struct snd_soc_jack hdmi_jack[3]; + +struct hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct card_private { + char codec_name[SND_ACPI_I2C_ID_LEN]; + struct snd_soc_jack headset; + struct list_head hdmi_pcm_list; +}; + +static const struct snd_kcontrol_new cml_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("WL Ext Spk"), + SOC_DAPM_PIN_SWITCH("WR Ext Spk"), +}; + +static const struct snd_kcontrol_new cml_rt1015_tt_controls[] = { + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + +static const struct snd_soc_dapm_widget cml_rt1015_rt5682_widgets[] = { + SND_SOC_DAPM_SPK("WL Ext Spk", NULL), + SND_SOC_DAPM_SPK("WR Ext Spk", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_widget cml_rt1015_tt_widgets[] = { + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cml_rt1015_rt5682_map[] = { + /*WL/WR speaker*/ + {"WL Ext Spk", NULL, "WL SPO"}, + {"WR Ext Spk", NULL, "WR SPO"}, + + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "IN1P", NULL, "Headset Mic" }, + + /* DMIC */ + { "DMic", NULL, "SoC DMIC" }, +}; + +static const struct snd_soc_dapm_route cml_rt1015_tt_map[] = { + /*TL/TR speaker*/ + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + +static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + /* need to enable ASRC function for 24MHz mclk rate */ + rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | + RT5682_AD_STEREO1_FILTER, + RT5682_CLK_SEL_I2S1_ASRC); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + + return ret; +}; + +static int cml_rt1015_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_card *card = rtd->card; + + if (sof_rt1015_quirk & (SOF_RT1015_SPEAKER_TL | + SOF_RT1015_SPEAKER_TR)) { + + ret = snd_soc_add_card_controls(card, cml_rt1015_tt_controls, + ARRAY_SIZE(cml_rt1015_tt_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, + cml_rt1015_tt_widgets, + ARRAY_SIZE(cml_rt1015_tt_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1015_tt_map, + ARRAY_SIZE(cml_rt1015_tt_map)); + + if (ret) + return ret; + } + + return ret; +} + +static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int clk_id, clk_freq, pll_out, ret; + + clk_id = RT5682_PLL1_S_MCLK; + clk_freq = CML_PLAT_CLK; + + pll_out = params_rate(params) * 512; + + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (ret < 0) + dev_warn(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret); + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_warn(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * slot_width should be equal or large than data length, set them + * be the same + */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, + params_width(params)); + if (ret < 0) + dev_warn(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; +} + +static int cml_rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + struct snd_soc_card *card = rtd->card; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + /* Set tdm/i2s1 master bclk ratio */ + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) { + dev_err(card->dev, "failed to set bclk ratio\n"); + return ret; + } + /* 64 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + 64 * srate, 256 * srate); + if (ret < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + RT1015_SCLK_S_PLL, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return ret; + } + + } + return 0; +} + +static struct snd_soc_ops cml_rt5682_ops = { + .hw_params = cml_rt5682_hw_params, +}; + +static const struct snd_soc_ops cml_rt1015_ops = { + .hw_params = cml_rt1015_hw_params, +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct hdmi_pcm *pcm; + int ret, i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + ret = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &hdmi_jack[i], + NULL, 0); + if (ret) + return ret; + + ret = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &hdmi_jack[i]); + if (ret < 0) + return ret; + + i++; + } + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +static int hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +/* Cometlake digital audio interface glue - connects codec <--> CPU */ + +SND_SOC_DAILINK_DEF(ssp0_pin, + DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); +SND_SOC_DAILINK_DEF(ssp0_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", + CML_RT5682_CODEC_DAI))); + +SND_SOC_DAILINK_DEF(ssp1_pin, + DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); +SND_SOC_DAILINK_DEF(ssp1_codec, + DAILINK_COMP_ARRAY( + /* WL */ COMP_CODEC("i2c-10EC1015:00", CML_RT1015_CODEC_DAI), + /* WR */ COMP_CODEC("i2c-10EC1015:01", CML_RT1015_CODEC_DAI))); + +SND_SOC_DAILINK_DEF(dmic_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); + +SND_SOC_DAILINK_DEF(dmic16k_pin, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); + +SND_SOC_DAILINK_DEF(dmic_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); + +SND_SOC_DAILINK_DEF(idisp1_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); +SND_SOC_DAILINK_DEF(idisp1_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); + +SND_SOC_DAILINK_DEF(idisp2_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); +SND_SOC_DAILINK_DEF(idisp2_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); + +SND_SOC_DAILINK_DEF(idisp3_pin, + DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); +SND_SOC_DAILINK_DEF(idisp3_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); + +static struct snd_soc_dai_link cml_rt1015_rt5682_dailink[] = { + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .init = cml_rt5682_codec_init, + .ignore_pmdown_time = 1, + .ops = &cml_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), + }, + { + .name = "dmic01", + .id = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), + }, + { + .name = "dmic16k", + .id = 2, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), + }, + { + .name = "iDisp1", + .id = 3, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), + }, + { + .name = "iDisp2", + .id = 4, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), + }, + { + .name = "iDisp3", + .id = 5, + .init = hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), + }, + { + /* + * SSP1 - Codec : added to end of list ensuring + * reuse of common topologies for other end points + * and changing only SSP1's codec + */ + .name = "SSP1-Codec", + .id = 6, + .dpcm_playback = 1, + .dpcm_capture = 1, /* Capture stream provides Feedback */ + .no_pcm = 1, + .init = cml_rt1015_spk_init, + .ops = &cml_rt1015_ops, + SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), + }, +}; + +static struct snd_soc_codec_conf rt1015_conf[] = { + { + .dev_name = "i2c-10EC1015:00", + .name_prefix = "WL", + }, + { + .dev_name = "i2c-10EC1015:01", + .name_prefix = "WR", + }, +}; + +/* Cometlake audio machine driver for RT1015 and RT5682 */ +static struct snd_soc_card snd_soc_card_cml = { + .name = "cml_rt1015_rt5682", + .dai_link = cml_rt1015_rt5682_dailink, + .num_links = ARRAY_SIZE(cml_rt1015_rt5682_dailink), + .codec_conf = rt1015_conf, + .num_configs = ARRAY_SIZE(rt1015_conf), + .dapm_widgets = cml_rt1015_rt5682_widgets, + .num_dapm_widgets = ARRAY_SIZE(cml_rt1015_rt5682_widgets), + .dapm_routes = cml_rt1015_rt5682_map, + .num_dapm_routes = ARRAY_SIZE(cml_rt1015_rt5682_map), + .controls = cml_controls, + .num_controls = ARRAY_SIZE(cml_controls), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static int snd_cml_rt1015_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link_component *rt1015_dais_components; + struct snd_soc_codec_conf *rt1015_dais_confs; + struct card_private *ctx; + struct snd_soc_acpi_mach *mach; + const char *platform_name; + int ret, i; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + mach = (&pdev->dev)->platform_data; + snd_soc_card_cml.dev = &pdev->dev; + platform_name = mach->mach_params.platform; + + dmi_check_system(sof_rt1015_quirk_table); + + dev_info(&pdev->dev, "sof_rt1015_quirk = %lx\n", sof_rt1015_quirk); + + if (sof_rt1015_quirk & (SOF_RT1015_SPEAKER_TL | + SOF_RT1015_SPEAKER_TR)) { + rt1015_dais_confs = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_codec_conf) * + SPK_CH, GFP_KERNEL); + + if (!rt1015_dais_confs) + return -ENOMEM; + + rt1015_dais_components = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link_component) * + SPK_CH, GFP_KERNEL); + + if (!rt1015_dais_components) + return -ENOMEM; + + for (i = 0; i < SPK_CH; i++) { + rt1015_dais_confs[i].dev_name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1015:0%d", + i); + + if (!rt1015_dais_confs[i].dev_name) + return -ENOMEM; + + switch (i) { + case 0: + rt1015_dais_confs[i].name_prefix = "WL"; + break; + case 1: + rt1015_dais_confs[i].name_prefix = "WR"; + break; + case 2: + rt1015_dais_confs[i].name_prefix = "TL"; + break; + case 3: + rt1015_dais_confs[i].name_prefix = "TR"; + break; + default: + return -EINVAL; + } + rt1015_dais_components[i].name = devm_kasprintf(&pdev->dev, + GFP_KERNEL, + "i2c-10EC1015:0%d", + i); + if (!rt1015_dais_components[i].name) + return -ENOMEM; + + rt1015_dais_components[i].dai_name = CML_RT1015_CODEC_DAI; + } + + snd_soc_card_cml.codec_conf = rt1015_dais_confs; + snd_soc_card_cml.num_configs = SPK_CH; + + for (i = 0; i < ARRAY_SIZE(cml_rt1015_rt5682_dailink); i++) { + if (!strcmp(cml_rt1015_rt5682_dailink[i].codecs->dai_name, + CML_RT1015_CODEC_DAI)) { + cml_rt1015_rt5682_dailink[i].codecs = rt1015_dais_components; + cml_rt1015_rt5682_dailink[i].num_codecs = SPK_CH; + } + } + } + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, + platform_name); + if (ret) + return ret; + snd_soc_card_set_drvdata(&snd_soc_card_cml, ctx); + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cml); +} + +static struct platform_driver snd_cml_rt1015_rt5682_driver = { + .probe = snd_cml_rt1015_probe, + .driver = { + .name = "cml_rt1015_rt5682", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(snd_cml_rt1015_rt5682_driver); + +/* Module information */ +MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1015 and RT5682 in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Shuming Fan <shumingf(a)realtek.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cml_rt1015_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index b80f032a..4f99da8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -23,6 +23,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_fw_filename = "sof-cnl.ri", .sof_tplg_filename = "sof-cnl-rt274.tplg", }, + { + .id = "10EC1015", + .drv_name = "cml_rt1015_rt5682", + .quirk_data = &cml_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-rt1015-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); -- 2.7.4

2 1

[PATCH] Asoc: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions
by Srinivasa Rao Mandadapu 26 Oct '20

26 Oct '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Update SC7180 lpass_variant structure with proper I2S bitwidth field bit positions, as bitwidth denotes 0 to 1 bits, but previously used only 0 bit. Fixes: commit cba62c8b49bead ("Merge series "ASoC: qcom: Add support for SC7180 lpass variant" from Rohit kumar <rohitkr(a)codeaurora.org>:") Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index c6292f9e..bc998d5 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = { .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH v3 28/56] slimbus: fix a kernel-doc markup
by Mauro Carvalho Chehab 26 Oct '20

26 Oct '20

Fix the name of the enum on its kernel-doc markup: enum slim_ch_aux_fmt -> enum slim_ch_aux_bit_fmt Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> --- drivers/slimbus/slimbus.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index c73035915f1d..00a7f112574b 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -244,7 +244,7 @@ enum slim_ch_data_fmt { }; /** - * enum slim_ch_aux_fmt: SLIMbus channel Aux Field format IDs according to + * enum slim_ch_aux_bit_fmt: SLIMbus channel Aux Field format IDs according to * Table 63 of SLIMbus Spec 2.0 * @SLIM_CH_AUX_FMT_NOT_APPLICABLE: Undefined * @SLIM_CH_AUX_FMT_ZCUV_TUNNEL_IEC60958: ZCUV for tunneling IEC60958 -- 2.26.2

2 1

[PATCH v3 48/56] sound: fix kernel-doc markups
by Mauro Carvalho Chehab 26 Oct '20

26 Oct '20

Kernel-doc markups should use this format: identifier - description There is a common comment marked, instead, with kernel-doc notation. Some identifiers have different names between their prototypes and the kernel-doc markup. Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> --- include/sound/core.h | 3 ++- include/sound/pcm.h | 4 ++-- include/uapi/sound/compress_offload.h | 2 +- sound/core/control.c | 4 ++-- sound/core/pcm_dmaengine.c | 3 ++- sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 4 ++-- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 2 +- 9 files changed, 14 insertions(+), 12 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 381a010a1bd4..0462c577d7a3 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -332,7 +332,8 @@ void __snd_printk(unsigned int level, const char *file, int line, #define snd_BUG() WARN(1, "BUG?\n") /** - * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. + * snd_printd_ratelimit - Suppress high rates of output when + * CONFIG_SND_DEBUG is enabled. */ #define snd_printd_ratelimit() printk_ratelimit() diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2ba5df2c9e23..2336bf9243e1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1284,8 +1284,8 @@ snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) } /** - * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig. - * page from the given size + * snd_pcm_sgbuf_get_chunk_size - Compute the max size that fits within the + * contig. page from the given size * @substream: PCM substream * @ofs: byte offset * @size: byte size to examine diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..9555f31c8425 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -144,7 +144,7 @@ struct snd_compr_metadata { __u32 value[8]; } __attribute__((packed, aligned(4))); -/** +/* * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec diff --git a/sound/core/control.c b/sound/core/control.c index 421ddc76f264..4373de42a5a0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1925,8 +1925,8 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl); #ifdef CONFIG_COMPAT /** - * snd_ctl_unregister_ioctl - de-register the device-specific compat 32bit - * control-ioctls + * snd_ctl_unregister_ioctl_compat - de-register the device-specific compat + * 32bit control-ioctls * @fcn: ioctl callback function to unregister */ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 4d059ff2b2e4..4d0e8fe535a1 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -356,7 +356,8 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); /** - * snd_dmaengine_pcm_release_chan_close - Close a dmaengine based PCM substream and release channel + * snd_dmaengine_pcm_close_release_chan - Close a dmaengine based PCM + * substream and release channel * @substream: PCM substream * * Releases the DMA channel associated with the PCM substream. diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d531e1bc2b81..bda3514c7b2d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -490,7 +490,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, EXPORT_SYMBOL(snd_pcm_set_ops); /** - * snd_pcm_sync - set the PCM sync id + * snd_pcm_set_sync - set the PCM sync id * @substream: the pcm substream * * Sets the PCM sync identifier for the card. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9e0b2d73faf6..47b155a49226 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -112,7 +112,7 @@ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); /** - * snd_pcm_stream_lock - Unlock the PCM stream + * snd_pcm_stream_unlock - Unlock the PCM stream * @substream: PCM substream * * This unlocks the PCM stream that has been locked via snd_pcm_stream_lock(). @@ -595,7 +595,7 @@ static void snd_pcm_sync_stop(struct snd_pcm_substream *substream) } /** - * snd_pcm_hw_param_choose - choose a configuration defined by @params + * snd_pcm_hw_params_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ea3986a46c12..05a085f6dc7c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2341,7 +2341,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, } /** - * snd_soc_unregister_dai - Unregister DAIs from the ASoC core + * snd_soc_unregister_dais - Unregister DAIs from the ASoC core * * @component: The component for which the DAIs should be unregistered */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 980f2c330b87..7f87b449f950 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1276,7 +1276,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, } /** - * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * snd_soc_dapm_dai_get_connected_widgets - query audio path and it's widgets. * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. -- 2.26.2

3 4

[PATCH] sound: remove unneeded break
by trix＠redhat.com 26 Oct '20

26 Oct '20

From: Tom Rix <trix(a)redhat.com> A break is not needed if it is preceded by a return, goto or break Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/pci/rme32.c | 1 - sound/pci/rme9652/hdspm.c | 8 -------- sound/pci/rme9652/rme9652.c | 7 ------- sound/soc/codecs/wcd-clsh-v2.c | 1 - sound/soc/codecs/wl1273.c | 1 - sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/ti/davinci-mcasp.c | 1 - 7 files changed, 20 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 869af8a32c98..4eabece4dcba 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -468,7 +468,6 @@ static int snd_rme32_capture_getrate(struct rme32 * rme32, int *is_adat) return 32000; default: return -1; - break; } else switch (n) { /* supporting the CS8412 */ diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4a1f576dd9cf..3382c069fd3d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2286,7 +2286,6 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 16) & 0xF; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); return (status >> HDSPM_AES32_wcFreq_bit) & 0xF; @@ -2312,7 +2311,6 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 20) & 0xF; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); return (status >> 1) & 0xF; @@ -2338,7 +2336,6 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_2); return (status >> 12) & 0xF; - break; default: break; } @@ -2358,7 +2355,6 @@ static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) case AES32: timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); return (timecode >> (4*index)) & 0xF; - break; default: break; } @@ -3845,7 +3841,6 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case MADI: status2 = hdspm_read(hdspm, HDSPM_statusRegister2); @@ -3856,7 +3851,6 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) return 1; } return 0; - break; case RayDAT: case AIO: @@ -3868,8 +3862,6 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) return 1; return 0; - break; - case MADIface: break; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 7ab10028d9fa..012fbec5e6a7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -732,34 +732,27 @@ static inline int rme9652_spdif_sample_rate(struct snd_rme9652 *s) switch (rme9652_decode_spdif_rate(rate_bits)) { case 0x7: return 32000; - break; case 0x6: return 44100; - break; case 0x5: return 48000; - break; case 0x4: return 88200; - break; case 0x3: return 96000; - break; case 0x0: return 64000; - break; default: dev_err(s->card->dev, "%s: unknown S/PDIF input rate (bits = 0x%x)\n", s->card_name, rate_bits); return 0; - break; } } diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c index 1be82113c59a..817d8259758c 100644 --- a/sound/soc/codecs/wcd-clsh-v2.c +++ b/sound/soc/codecs/wcd-clsh-v2.c @@ -480,7 +480,6 @@ static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, case WCD_CLSH_STATE_HPHR: wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); break; - break; case WCD_CLSH_STATE_LO: wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); break; diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index c56b9329240f..d8ced4559bf2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -311,7 +311,6 @@ static int wl1273_startup(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } return 0; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index bbe8d782e0af..b1ca64d2f7ea 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -502,7 +502,6 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) return ret; return skl_run_pipe(skl, mconfig->pipe); - break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a6b72ad53b43..2d85cc4c67fb 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2385,7 +2385,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); case -EPROBE_DEFER: goto err; - break; } if (ret) { -- 2.18.1

3 2

DOC or BUG: plugins dsnoop, dmix and dshare change mode parameter of snd_pcm_open_slave
by GitHub issues - opened 26 Oct '20

26 Oct '20

alsa-project/alsa-lib issue #92 was opened from RonaldAJ: Using the dsnoop plugin sets the mode to non-blocking in alsa-lib/src/pcm/pcm_dsnoop.c: ret = snd_pcm_open_slave(&spcm, root, sconf, stream, mode | SND_PCM_NONBLOCK | SND_PCM_APPEND, NULL); Judging from the code the same applies for dmix, dshare. It took me a considerable amount of time to figure this out using the source code. So it would be nice if this behaviour where documented. Personally I would prefer the code to throw an error message, when using blocking mode with these plugins. But I don't know how that ties in with the overall design. Issue URL : https://github.com/alsa-project/alsa-lib/issues/92 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RESEND v2] ASoC: tegra20-spdif: remove "default m"
by Michał Mirosław 26 Oct '20

26 Oct '20

Make tegra20-spdif default to N as all other drivers do. Add the selection to defconfigs instead. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Fixes: 774fec338bfc ("ASoC: Tegra: Implement SPDIF CPU DAI") --- v2: add the symbol to defconfig as suggested by Thierry Reding --- arch/arm/configs/multi_v7_defconfig | 1 + arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 1 - 3 files changed, 2 insertions(+), 1 deletion(-) diff --git a/arch/arm/configs/multi_v7_defconfig b/arch/arm/configs/multi_v7_defconfig index e9e76e32f10f..19342ac738a5 100644 --- a/arch/arm/configs/multi_v7_defconfig +++ b/arch/arm/configs/multi_v7_defconfig @@ -743,6 +743,7 @@ CONFIG_SND_SOC_STM32_I2S=m CONFIG_SND_SUN4I_CODEC=m CONFIG_SND_SOC_TEGRA=m CONFIG_SND_SOC_TEGRA20_I2S=m +CONFIG_SND_SOC_TEGRA20_SPDIF=m CONFIG_SND_SOC_TEGRA30_I2S=m CONFIG_SND_SOC_TEGRA_RT5640=m CONFIG_SND_SOC_TEGRA_WM8753=m diff --git a/arch/arm/configs/tegra_defconfig b/arch/arm/configs/tegra_defconfig index fff5fae0db30..08526eb50484 100644 --- a/arch/arm/configs/tegra_defconfig +++ b/arch/arm/configs/tegra_defconfig @@ -225,6 +225,7 @@ CONFIG_SND_HDA_CODEC_HDMI=y CONFIG_SND_SOC=y CONFIG_SND_SOC_TEGRA=y CONFIG_SND_SOC_TEGRA20_I2S=y +CONFIG_SND_SOC_TEGRA20_SPDIF=y CONFIG_SND_SOC_TEGRA30_I2S=y CONFIG_SND_SOC_TEGRA_RT5640=y CONFIG_SND_SOC_TEGRA_WM8753=y diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 3d91bd3e59cd..a62cc87551ac 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -39,7 +39,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support -- 2.20.1

2 1

[PATCH mlx5-next 00/11] Convert mlx5 to use auxiliary bus
by Leon Romanovsky 26 Oct '20

26 Oct '20

From: Leon Romanovsky <leonro(a)nvidia.com> Hi, This patch set converts mlx5 driver to use auxiliary bus [1]. In this series, we are connecting three subsystems (VDPA, netdev and RDMA) through mlx5_core PCI driver. That driver is responsible to create proper devices based on supported firmware. First four patches are preparitions and fixes that were spotted during code development, rest is the conversion itself. Thanks [1] https://lore.kernel.org/lkml/20201023003338.1285642-1-david.m.ertman@intel.… Leon Romanovsky (11): net/mlx5: Don't skip vport check net/mlx5: Properly convey driver version to firmware net/mlx5_core: Clean driver version and name vdpa/mlx5: Make hardware definitions visible to all mlx5 devices net/mlx5: Register mlx5 devices to auxiliary virtual bus vdpa/mlx5: Connect mlx5_vdpa to auxiliary bus net/mlx5e: Connect ethernet part to auxiliary bus RDMA/mlx5: Convert mlx5_ib to use auxiliary bus net/mlx5: Delete custom device management logic net/mlx5: Simplify eswitch mode check RDMA/mlx5: Remove IB representors dead code drivers/infiniband/hw/mlx5/counters.c | 7 - drivers/infiniband/hw/mlx5/ib_rep.c | 113 ++-- drivers/infiniband/hw/mlx5/ib_rep.h | 45 +- drivers/infiniband/hw/mlx5/main.c | 148 +++-- drivers/infiniband/hw/mlx5/mlx5_ib.h | 4 +- .../net/ethernet/mellanox/mlx5/core/Kconfig | 1 + drivers/net/ethernet/mellanox/mlx5/core/dev.c | 504 ++++++++++-------- .../net/ethernet/mellanox/mlx5/core/devlink.c | 4 +- .../ethernet/mellanox/mlx5/core/en_ethtool.c | 4 +- .../net/ethernet/mellanox/mlx5/core/en_main.c | 135 ++--- .../net/ethernet/mellanox/mlx5/core/en_rep.c | 42 +- .../net/ethernet/mellanox/mlx5/core/en_rep.h | 6 +- .../net/ethernet/mellanox/mlx5/core/en_tc.c | 8 +- .../mellanox/mlx5/core/esw/devlink_port.c | 2 +- .../net/ethernet/mellanox/mlx5/core/eswitch.c | 28 +- .../mellanox/mlx5/core/eswitch_offloads.c | 6 + .../mellanox/mlx5/core/ipoib/ethtool.c | 2 +- drivers/net/ethernet/mellanox/mlx5/core/lag.c | 58 +- .../net/ethernet/mellanox/mlx5/core/main.c | 43 +- .../ethernet/mellanox/mlx5/core/mlx5_core.h | 28 +- drivers/vdpa/mlx5/Makefile | 2 +- drivers/vdpa/mlx5/net/main.c | 76 --- drivers/vdpa/mlx5/net/mlx5_vnet.c | 55 +- drivers/vdpa/mlx5/net/mlx5_vnet.h | 24 - include/linux/mlx5/driver.h | 36 +- include/linux/mlx5/eswitch.h | 8 +- .../linux/mlx5/mlx5_ifc_vdpa.h | 6 +- 27 files changed, 731 insertions(+), 664 deletions(-) delete mode 100644 drivers/vdpa/mlx5/net/main.c delete mode 100644 drivers/vdpa/mlx5/net/mlx5_vnet.h rename drivers/vdpa/mlx5/core/mlx5_vdpa_ifc.h => include/linux/mlx5/mlx5_ifc_vdpa.h (97%) -- 2.26.2

1 11

[PATCH 1/2] control: Add documentation for snd_ctl_card_* and friends.
by Tanjeff-N. Moos 25 Oct '20

25 Oct '20

In this patch series, I added a description about control interface handling and how control interfaces are identified. In addition, I added/improved Doxygen documentation for the snd_ctl_card_info_t type and related corresponding functions, e.g. snd_ctl_card_info(). I also documented other card-related like snd_card_next(). Along the way I did minor documentation improvements. Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- include/control.h | 40 +++++++++- src/control/cards.c | 56 +++++++++----- src/control/control.c | 175 +++++++++++++++++++++++++++++++----------- 3 files changed, 202 insertions(+), 69 deletions(-) diff --git a/include/control.h b/include/control.h index 8766f440..e1a0da47 100644 --- a/include/control.h +++ b/include/control.h @@ -50,7 +50,32 @@ typedef struct snd_aes_iec958 { unsigned char dig_subframe[4]; /**< AES/IEC958 subframe bits */ } snd_aes_iec958_t; -/** CTL card info container */ +/** \brief CTL card info container. + * + * This type contains meta information about a sound card, such as the index, + * name, longname, etc. + * + * \par Memory management + * + * Before using a snd_ctl_card_info_t object, it must be allocated using + * snd_ctl_card_info_alloca() or snd_ctl_card_info_malloc(). When using the + * latter, it must be freed again using snd_ctl_card_info_free(). + * + * A card info object can be zeroed out using snd_ctl_card_info_clear(). + * + * A card info object can be copied to another one using + * snd_ctl_card_info_copy(). + * + * \par Obtaining the Information + * + * To obtain the card information, it must first be opened using + * snd_ctl_open(), and a snd_ctl_card_info_t container must be + * allocated. Then, the information can be read using + * snd_ctl_card_info_get_card(). + * + * Thereafter, the card properties can be read using the + * snd_ctl_card_info_get_*() functions. + */ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ @@ -438,11 +463,20 @@ void snd_ctl_elem_id_set_name(snd_ctl_elem_id_t *obj, const char *val); void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val); size_t snd_ctl_card_info_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_card_info_t using standard alloca - * \param ptr returned pointer + * \brief Allocate an invalid #snd_ctl_card_info_t on the stack. + * + * Allocate space for a card info object on the stack. The allocated + * memory need not be freed, because it is on the stack. + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_elem_value_t pointer. The address + * of the allocated space will returned here. */ #define snd_ctl_card_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_card_info) + int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr); void snd_ctl_card_info_free(snd_ctl_card_info_t *obj); void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj); diff --git a/src/control/cards.c b/src/control/cards.c index e57457c2..8226c42d 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -77,8 +77,8 @@ static int snd_card_load1(int card) /** * \brief Try to load the driver for a card. - * \param card Card number. - * \return 1 if driver is present, zero if driver is not present + * \param card Card index. + * \return 1 if driver is present, zero if driver is not present. */ int snd_card_load(int card) { @@ -86,14 +86,24 @@ int snd_card_load(int card) } /** - * \brief Try to determine the next card. - * \param rcard pointer to card number - * \result zero if success, otherwise a negative error code + * \brief Iterate over physical sound cards. + * + * This function takes the index of a physical sound card and sets it to the + * index of the next card. If index is -1, it is set to the index of the first + * card. After the last card, the index is set to -1. + * + * For example, if you have 2 sound cards (with index 0 and 1), the index will + * be modified as follows: + * + * - -1 --> 0 + * - 0 --> 1 + * - 1 --> -1 + * + * This does not work for virtual sound cards. * - * Tries to determine the next card from given card number. - * If card number is -1, then the first available card is - * returned. If the result card number is -1, no more cards - * are available. + * \param rcard Index of current card. The index of the next card is stored + * here. + * \result zero if success, otherwise a negative error code. */ int snd_card_next(int *rcard) { @@ -114,13 +124,18 @@ int snd_card_next(int *rcard) } /** - * \brief Convert card string to an integer value. - * \param string String containing card identifier - * \return zero if success, otherwise a negative error code + * \brief Convert a card string to the card index. * - * The accepted format is an integer value in ASCII representation - * or the card identifier (the id parameter for sound-card drivers). - * The control device name like /dev/snd/controlC0 is accepted, too. + * This works only for physical sound cards, not for virtual cards. + * + * \param string A string identifying the card. + * \return The index of the card. On error, a a negative error code + * is returned. + * + * The accepted formats for "string" are: + * - The index of the card (as listed in /proc/asound/cards), given as string + * - The ID of the card (as listed in /proc/asound/cards) + * - The control device name (like /dev/snd/controlC0) */ int snd_card_get_index(const char *string) { @@ -163,8 +178,9 @@ int snd_card_get_index(const char *string) /** * \brief Obtain the card name. - * \param card Card number - * \param name Result - card name corresponding to card number + * + * \param card The index of the card. + * \param name Result - card name corresponding to card index. * \result zero if success, otherwise a negative error code * * The value returned in name is allocated with strdup and should be @@ -193,9 +209,9 @@ int snd_card_get_name(int card, char **name) /** * \brief Obtain the card long name. - * \param card Card number - * \param name Result - card long name corresponding to card number - * \result zero if success, otherwise a negative error code + * \param card Index of the card. + * \param name Result - card long name corresponding to card index. + * \result Zero if success, otherwise a negative error code. * * The value returned in name is allocated with strdup and should be * freed when no longer used. diff --git a/src/control/control.c b/src/control/control.c index 08058c06..7abf8ee3 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -33,8 +33,50 @@ <P>Control interface is designed to access primitive controls. There is also an interface for notifying about control and structure changes. + \section control_general_overview General overview +In Alsa, there are physical sound cards, such as USB headsets, and +virtual sound cards, such as "pulse", which represents the PulseAudio +Sound system. Each sound card offers a control interface, making its +settings (e.g. volume knobs) available. The complete list of available +control interfaces can be obtained using snd_device_name_hint(), +giving -1 as card index and "ctl" as interface type. Each returned +NAME hint identifies a control interface. + +Sound cards have an ID (a string), an index (an int, sometimes called +the "card number"), a name, a longname, a mixername and a "components" +property. The file /proc/asound/cards lists most of these properties +for physical sound cards. Virtual sound cards are not listed in that +file. The format is: + +\verbatim +index [ID ] Driver - name + longname +\endverbatim + +Note that the mixername and components are not listed. + + +\subsection control_cards_id Identifying and Opening Control Interfaces + +To work with a control interface, is must be opened first, using +snd_ctl_open(). This function takes the interface name. + +For physical sound cards, the control interface can be identified +using the string "hw:<index>" (e.g. `hw:2`). The NAME hint - which is +"hw:CARD=<ID>" - can also be used. Further, its device file (something +like `/dev/snd/controlC0`) is also acceptable. Either of them can be +given to snd_ctl_open(). + +For virtual sound cards, the NAME hint is given to snd_ctl_open(). + +The functions snd_card_get_index(), snd_card_get_name() and +snd_card_get_longname() can be used to find an identifying property +when another one is already known. + +\section control_elements Elements + In ALSA control feature, each sound card can have control elements. The elements are managed according to below model. @@ -65,7 +107,7 @@ are managed according to below model. of userspace applications and drivers in kernel. -\section identifying_elements Identifying Elements +\subsection identifying_elements Identifying Elements Each element has the following identifying properties: @@ -84,7 +126,7 @@ rare). The numid can change on each boot. In case of an USB sound card, the numid can also change when it is reconnected. -\section element_lists Element Lists +\subsection element_lists Element Lists An element list can be used to obtain a list of all elements of the sound card. The list contains generic information (e.g. how many @@ -93,7 +135,7 @@ elements the card has), and the identifying properties of the elements element lists. -\section working_with_elements Working with Elements +\subsection working_with_elements Working with Elements It is possible to obtain information about an element using the snd_ctl_elem_info_*() functions. For enums, the allowed values can be @@ -108,7 +150,7 @@ actual values or settings. It is also possible to get and set the ID values (such as the numid or the name). -\section element_sets Element Sets +\subsection element_sets Element Sets The type of element set is one of integer, integer64, boolean, enumerators, bytes and IEC958 structure. This indicates the type of value for each member in @@ -328,10 +370,15 @@ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe) /** - * \brief Get card related information - * \param ctl CTL handle - * \param info Card info pointer - * \return 0 on success otherwise a negative error code + * \brief Get information about the sound card. + * + * Obtain information about the sound card previously opened using + * snd_ctl_open(). The object "info" must be allocated prior to calling this + * function. See snd_ctl_card_info_t for details. + * + * \param ctl The CTL handle. + * \param info The card information is stored here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) { @@ -1477,11 +1524,13 @@ static int snd_ctl_open_noupdate(snd_ctl_t **ctlp, snd_config_t *root, const cha } /** - * \brief Opens a CTL - * \param ctlp Returned CTL handle - * \param name ASCII identifier of the CTL handle - * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC) - * \return 0 on success otherwise a negative error code + * \brief Opens a sound card. + * + * \param ctlp Returned CTL handle. + * \param name A string identifying the card (See \ref control_cards_id). + * \param mode Open mode (see #SND_CTL_NONBLOCK, #SND_CTL_ASYNC). + * + * \return 0 on success otherwise a negative error code. */ int snd_ctl_open(snd_ctl_t **ctlp, const char *name, int mode) { @@ -1925,8 +1974,8 @@ void snd_ctl_elem_id_set_index(snd_ctl_elem_id_t *obj, unsigned int val) } /** - * \brief get size of #snd_ctl_card_info_t - * \return size in bytes + * \brief get size of #snd_ctl_card_info_t. + * \return Size in bytes. */ size_t snd_ctl_card_info_sizeof() { @@ -1934,9 +1983,16 @@ size_t snd_ctl_card_info_sizeof() } /** - * \brief allocate an invalid #snd_ctl_card_info_t using standard malloc - * \param ptr returned pointer - * \return 0 on success otherwise negative error code + * \brief Allocate an invalid #snd_ctl_card_info_t on the heap. + * + * Allocate space for a card info object on the heap. The allocated memory + * must be freed using snd_ctl_card_info_free(). + * + * See snd_ctl_card_info_t for details. + * + * \param ptr Pointer to a snd_ctl_card_info_t pointer. The address + * of the allocated space will be returned here. + * \return 0 on success, otherwise a negative error code. */ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) { @@ -1948,8 +2004,10 @@ int snd_ctl_card_info_malloc(snd_ctl_card_info_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_card_info_t - * \param obj pointer to object to free + * \brief Free an #snd_ctl_card_info_t previously allocated using + * snd_ctl_card_info_malloc(). + * + * \param obj Pointer to the snd_ctl_card_info_t. */ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) { @@ -1957,8 +2015,11 @@ void snd_ctl_card_info_free(snd_ctl_card_info_t *obj) } /** - * \brief clear given #snd_ctl_card_info_t object - * \param obj pointer to object to clear + * \brief Clear given card info object. + * + * See snd_ctl_elem_value_t for details. + * + * \param obj Card info object. */ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) { @@ -1966,9 +2027,10 @@ void snd_ctl_card_info_clear(snd_ctl_card_info_t *obj) } /** - * \brief copy one #snd_ctl_card_info_t to another - * \param dst pointer to destination - * \param src pointer to source + * \brief Bitwise copy of a #snd_ctl_card_info_t object. + * + * \param dst Pointer to destination. + * \param src Pointer to source. */ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t *src) { @@ -1977,9 +2039,12 @@ void snd_ctl_card_info_copy(snd_ctl_card_info_t *dst, const snd_ctl_card_info_t } /** - * \brief Get card number from a CTL card info - * \param obj CTL card info - * \return card number + * \brief Get the sound card index from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card index. */ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) { @@ -1988,9 +2053,12 @@ int snd_ctl_card_info_get_card(const snd_ctl_card_info_t *obj) } /** - * \brief Get card identifier from a CTL card info - * \param obj CTL card info - * \return card identifier + * \brief Get the sound card ID from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card ID. */ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) { @@ -1999,9 +2067,12 @@ const char *snd_ctl_card_info_get_id(const snd_ctl_card_info_t *obj) } /** - * \brief Get card driver name from a CTL card info - * \param obj CTL card info - * \return card driver name + * \brief Get the sound card driver from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return The sound card driver. */ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) { @@ -2010,9 +2081,12 @@ const char *snd_ctl_card_info_get_driver(const snd_ctl_card_info_t *obj) } /** - * \brief Get card name from a CTL card info - * \param obj CTL card info - * \return card name + * \brief Get the sound card name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card name. */ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) { @@ -2021,9 +2095,12 @@ const char *snd_ctl_card_info_get_name(const snd_ctl_card_info_t *obj) } /** - * \brief Get card long name from a CTL card info - * \param obj CTL card info - * \return card long name + * \brief Get the sound cards long name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards long name. */ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) { @@ -2032,9 +2109,12 @@ const char *snd_ctl_card_info_get_longname(const snd_ctl_card_info_t *obj) } /** - * \brief Get card mixer name from a CTL card info - * \param obj CTL card info - * \return card mixer name + * \brief Get the sound card mixer name from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound card mixer name. */ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) { @@ -2043,9 +2123,12 @@ const char *snd_ctl_card_info_get_mixername(const snd_ctl_card_info_t *obj) } /** - * \brief Get card component list from a CTL card info - * \param obj CTL card info - * \return card mixer identifier + * \brief Get the sound cards "components" property from the given info object. + * + * See snd_ctl_card_info_t for more details. + * + * \param obj The card info object. + * \return Sound cards "components" property. */ const char *snd_ctl_card_info_get_components(const snd_ctl_card_info_t *obj) { -- 2.17.1

1 1

[PATCH] ASoC: codecs: wsa881x: add missing stream rates and format
by Srinivas Kandagatla 24 Oct '20

24 Oct '20

Add missing supported rates and formats for the stream, without which attempt to do playback will fail to find any matching rates/format. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa881x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index d39d479e2378..5456124457a7 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = { .id = 0, .playback = { .stream_name = "SPKR Playback", + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_max = 48000, .rate_min = 48000, .channels_min = 1, -- 2.21.0

2 1

[PATCH v2] ASoC: cs42l51: manage mclk shutdown delay
by Olivier Moysan 24 Oct '20

24 Oct '20

A delay must be introduced before the shutdown down of the mclk, as stated in CS42L51 datasheet. Otherwise the codec may produce some noise after the end of DAPM power down sequence. The delay between DAC and CLOCK_SUPPLY widgets is too short. Add a delay in mclk shutdown request to manage the shutdown delay explicitly. From experiments, at least 10ms delay is necessary. Set delay to 20ms as recommended in Documentation/timers/timers-howto.rst when using msleep(). Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Recommended Power-Down Sequence: (see https://statics.cirrus.com/pubs/proDatasheet/CS42L51_F2.pdf) 1. Mute the DACs and ADCs. 2. Disable soft ramp and zero-cross volume transitions 3. Set the PDN bit to 1. 4. Wait at least 100 μs. The codec will be fully powered down after this 100 μs delay. Prior to the removal of the master clock(MCLK), this delay of at least 100 μs must be implemented after step 3 to avoid premature disruption of the codec’s power down sequence. Changes in v2: - Manage explicitly DAPM events through a switch case in mclk_event() --- sound/soc/codecs/cs42l51.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 097c4e8d9950..c61b17dc2af8 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(cs42l51->mclk_handle); + case SND_SOC_DAPM_POST_PMD: + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { -- 2.17.1

2 1

[PATCH] ASoC: SOF: loader: handle all SOF_IPC_EXT types
by Kai Vehmanen 24 Oct '20

24 Oct '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Do not emit a warning for extended firmware header fields that are not used by kernel. This creates unnecessary noise to kernel logs like: sof-audio-pci 0000:00:1f.3: warning: unknown ext header type 3 size 0x1c sof-audio-pci 0000:00:1f.3: warning: unknown ext header type 4 size 0x10 Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/loader.c | 5 +++++ 1 file changed, 5 insertions(+) - baseline: eb5a558705c7 commit ("ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015") diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 68ed454f7ddf..ba9ed66f98bc 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) case SOF_IPC_EXT_CC_INFO: ret = get_cc_info(sdev, ext_hdr); break; + case SOF_IPC_EXT_UNUSED: + case SOF_IPC_EXT_PROBE_INFO: + case SOF_IPC_EXT_USER_ABI_INFO: + /* They are supported but we don't do anything here */ + break; default: dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); -- 2.28.0

2 1

[PATCH] ASoC: qcom: sdm845: set driver name correctly
by Srinivas Kandagatla 24 Oct '20

24 Oct '20

With the current state of code, we would endup with something like below in /proc/asound/cards for 2 machines based on this driver. Machine 1: 0 [DB845c ]: DB845c - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: Lenovo-YOGA-C63 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 This is not very UCM friendly both w.r.t to common up configs and card identification, and UCM2 became totally not usefull with just one ucm sdm845.conf for two machines which have different setups w.r.t HDMI and other dais. Reasons for such thing is partly because Qualcomm machine drivers never cared to set driver_name. This patch sets up driver name for the this driver to sort out the UCM integration issues! after this patch contents of /proc/asound/cards: Machine 1: 0 [DB845c ]: sdm845 - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: sdm845 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 with this its possible to align with what UCM2 expects and we can have sdm845/DB845.conf sdm845/LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216.conf ... for board variants. This should scale much better! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I will post the ucm changes once this patch is merged! sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index ab1bf23c21a6..6c2760e27ea6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -17,6 +17,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) if (!data) return -ENOMEM; + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; -- 2.21.0

2 1

[PATCH 0/4] ASoC: SOF: cleanups for 5.10
by Kai Vehmanen 24 Oct '20

24 Oct '20

Series with multiple code cleanups, plus one fix to remove unnecessary kernel warnings related to firmware loading. Bard Liao (1): ASoC: SOF: loader: handle all SOF_IPC_EXT types Pierre-Louis Bossart (3): ASoC: SOF: control: remove const in sizeof() ASoC: SOF: topology: remove const in sizeof() ASoC: SOF: sof-audio: remove goto used for force-nocodec support sound/soc/sof/control.c | 20 ++++++++++---------- sound/soc/sof/loader.c | 5 +++++ sound/soc/sof/sof-audio.c | 10 +++------- sound/soc/sof/topology.c | 2 +- 4 files changed, 19 insertions(+), 18 deletions(-) -- 2.27.0

3 6

[PATCH 1/4] ALSA: hda: Refactor codec PM to use direct-complete optimization
by Kai-Heng Feng 23 Oct '20

23 Oct '20

Upon system resume, hda_codec_pm_resume() uses hda_codec_force_resume() to resume the codec. However, pm_runtime_force_resume() won't really resume the codec because of pm_runtime_need_not_resume() check. Hence, hda_codec_force_resume() schedules a jackpoll work, which is to really power up the codec. Instead of doing that, we can use direct-complete to make the PM flow more straightforward, and keep codec always suspended through system PM flow if conditions are met. On system suspend, PM core will decide what to do based on hda_codec_pm_prepare(): - If codec is not runtime-suspended, PM core will suspend and resume the device as normal. - If codec is runtime-suspended, PM core will try to keep it suspended. If it's still suspended after system resume, we use hda_codec_pm_complete() to resume codec if it's needed. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_codec.c | 45 +++++++++++++++++++++++++-------------- 1 file changed, 29 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a356c21edb90..c2a510cc81bb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2934,7 +2934,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hdac_leave_pm(&codec->core); } -static int hda_codec_runtime_suspend(struct device *dev) +static int hda_codec_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); unsigned int state; @@ -2953,7 +2953,7 @@ static int hda_codec_runtime_suspend(struct device *dev) return 0; } -static int hda_codec_runtime_resume(struct device *dev) +static int hda_codec_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -2967,57 +2967,70 @@ static int hda_codec_runtime_resume(struct device *dev) pm_runtime_mark_last_busy(dev); return 0; } + +static int hda_codec_runtime_suspend(struct device *dev) +{ + return hda_codec_suspend(dev); +} + +static int hda_codec_runtime_resume(struct device *dev) +{ + return hda_codec_resume(dev); +} + #endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP -static int hda_codec_force_resume(struct device *dev) +static int hda_codec_pm_prepare(struct device *dev) +{ + return pm_runtime_suspended(dev); +} + +static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - int ret; - ret = pm_runtime_force_resume(dev); - /* schedule jackpoll work for jack detection update */ - if (codec->jackpoll_interval || - (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); - return ret; + if (pm_runtime_suspended(dev) && + (hda_codec_need_resume(codec) || codec->forced_resume)) + pm_request_resume(dev); } static int hda_codec_pm_suspend(struct device *dev) { dev->power.power_state = PMSG_SUSPEND; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_freeze(struct device *dev) { dev->power.power_state = PMSG_FREEZE; - return pm_runtime_force_suspend(dev); + return hda_codec_suspend(dev); } static int hda_codec_pm_thaw(struct device *dev) { dev->power.power_state = PMSG_THAW; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } static int hda_codec_pm_restore(struct device *dev) { dev->power.power_state = PMSG_RESTORE; - return hda_codec_force_resume(dev); + return hda_codec_resume(dev); } #endif /* CONFIG_PM_SLEEP */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { #ifdef CONFIG_PM_SLEEP + .prepare = hda_codec_pm_prepare, + .complete = hda_codec_pm_complete, .suspend = hda_codec_pm_suspend, .resume = hda_codec_pm_resume, .freeze = hda_codec_pm_freeze, -- 2.17.1

3 12

[PATCH] ASoC: Intel: Enable coredump in kconfig
by Gustaw Lewandowski 23 Oct '20

23 Oct '20

From: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> WANT_DEV_COREDUMP flag is needed here if this driver is only one using dev_coredump feature. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a5b446d5af19..c215156e2e18 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -4,6 +4,7 @@ config SND_SOC_INTEL_SST_TOPLEVEL default y depends on X86 || COMPILE_TEST select SND_SOC_INTEL_MACH + select WANT_DEV_COREDUMP help Intel ASoC SST Platform Drivers. If you have a Intel machine that has an audio controller with a DSP and I2S or DMIC port, then -- 2.29.0

2 1

Re: [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.2.4-1-g0c5948e
by Takashi Iwai 23 Oct '20

23 Oct '20

On Fri, 23 Oct 2020 11:41:25 +0200, GIT server wrote: > > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "ALSA utilities repository". > > The branch, master has been updated > via 0c5948e98a6a8535c89b7bcab13017d7732181c6 (commit) > from b2ae0b074669f976c53a52bcd0129227321f88c9 (commit) > > Those revisions listed above that are new to this repository have > not appeared on any other notification email; so we list those > revisions in full, below. > > - Log ----------------------------------------------------------------- > commit 0c5948e98a6a8535c89b7bcab13017d7732181c6 > Author: Hui Wang <hui.wang(a)canonical.com> > Date: Fri Oct 23 16:47:10 2020 +0800 > > aplay: try to use 16-bit format to increase capture quality > > Recently users reported a bug, I tested it and found it is a common > issue on Laptop or Desktop machines. > > The issue is users plug a headset and use "arecord test.wav" to > record a sound with default input volume, the recorded sound has > poor quality and nearly can't distinguish it is the sound we want > to record. > > This is because the input volume is low and the default format is U8. > The driver records sound with 16bit, because the input volume is low, > most of samples are within (-256,+256), when converting 16bit to U8, > those samples will be 0x7f. This is called quantization noise and we > could only workaround it by increase the input volume or adding -f to > arecord. > > But users want to record a better quality sound with default input > volume (after installing a new OS, the volume is the default volume), > and they don't want to add parameters to the arecord because most of > new linux users just use "arecord test.wav". > > So this patch tries to change the default format from U8 to S16_LE/BE. > If the machine doesn't support S16_LE/BE, it still uses U8 as default > format. > > Signed-off-by: Hui Wang <hui.wang(a)canonical.com> > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> This brought a slight concern about the compatibility, as the tool is very old and used in many places. But a right usage is always with the proper option, so maybe this won't break too many things (hopefully). Though, this change made me wonder why we don't have any default configuration for aplay/arecord. With that, you can adjust it per user or per distro. Would it bring more benefit? thanks, Takashi

1 0

PA - snd_pcm_avail() returned a value that is exceptionally large
by GitHub issues - edited 23 Oct '20

23 Oct '20

alsa-project/alsa-lib issue #68 was edited from reezer: I hope this is the right place to report this. This occurred in relation to a crash. http://alsa-project.org/db/?f=52e8505d884682fc06adac4ab8ce92a4ca3e0870 ``` E: [alsa-sink-ALC887-VD Analog] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 177312 bytes (923 ms). E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: [alsa-sink-ALC887-VD Analog] alsa-util.c: snd_pcm_dump(): E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Soft volume PCM E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Control: PCM Playback Volume E: [alsa-sink-ALC887-VD Analog] alsa-util.c: min_dB: -51 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: max_dB: 0 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: resolution: 256 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: Its setup is: E: [alsa-sink-ALC887-VD Analog] alsa-util.c: stream : PLAYBACK E: [alsa-sink-ALC887-VD Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-sink-ALC887-VD Analog] alsa-util.c: format : S16_LE E: [alsa-sink-ALC887-VD Analog] alsa-util.c: subformat : STD E: [alsa-sink-ALC887-VD Analog] alsa-util.c: channels : 2 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: rate : 48000 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-sink-ALC887-VD Analog] alsa-util.c: msbits : 16 E: [alsa-sink-ALC887-VD Analog] alsa-util.c: buffer_size : 4800 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/68 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

PA poll wakeup issue (was: bug in the ALSA driver 'snd_hda_intel')
by GitHub issues - edited 23 Oct '20

23 Oct '20

alsa-project/alsa-lib issue #90 was edited from Mysgym: Hello, after spending some time troubleshooting an issue with an audio interface (using snd_hda_intel). I've stumbled upon this message from PulseAudio in the journal : `Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ` So I filed a bug report This is Manjaro on Kernel 5.8 Issue URL : https://github.com/alsa-project/alsa-lib/issues/90 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

aplay: try to use 16-bit format to increase capture quality
by GitHub pull_request - opened 23 Oct '20

23 Oct '20

alsa-project/alsa-utils pull request #69 was opened from jason77-wang: Recently users reported a bug, I tested it and found it is a common issue on Laptop or Desktop machines. The issue is users plug a headset and use "arecord test.wav" to record a sound with default input volume, the recorded sound has poor quality and nearly can't distinguish it is the sound we want to record. This is because the input volume is low and the default format is U8. The driver records sound with 16bit, because the input volume is low, most of samples are within (-256,+256), when converting 16bit to U8, those samples will be 0x7f. This is called quantization noise and we could only workaround it by increase the input volume or adding -f to arecord. But users want to record a better quality sound with default input volume (after installing a new OS, the volume is the default volume), and they don't want to add parameters to the arecord because most of new linux users just use "arecord test.wav". So this patch tries to change the default format from U8 to S16_LE/BE. If the machine doesn't support S16_LE/BE, it still uses U8 as default format. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/69 Patch URL : https://github.com/alsa-project/alsa-utils/pull/69.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel mailing list / gmail] Re: Excessive bounces
by Jaroslav Kysela 23 Oct '20

23 Oct '20

Dne 21. 10. 20 v 23:09 Eu, acc napsal(a): > Dear Sir > > I received an email to confirm membership on alsa-devel list. > > I have doubled checked configuration, possible invasion of my systems. It is > strange because I use Gmail for reading the list, there is more than 5 GB of > available space, and not other reason for the bounces. > > The only curious point is that email from several regular developers, even > Jaroslav at a time, are classified as spam by gmail. I tried to discover the > reason in the past, but the explanation from Google is that there seems to be > a lot of spam in alada-devel and and alsa-users lists. I need the real examples (for alsa-devel). The alsa-users mailing list is on sourceforge and probably dead. > Thanks and keep the good work. [Cc: to alsa-devel] I'm sorry, but it seems that gmail changed something before several weeks for incoming e-mail and they have weird message rate limit rules for IP address and content classification. It basically does not work for the mailing list. I already tried to contact their support, but I've not received any answer. I already implemented all protections like DKIM, DMARC, SFP. Verified today using a dmarc report analyzer tool. Unfortunately, I cannot do much here. I can assist to fix the issues, but I need to know, what I can improve. I suggest to use another e-mail service. It seems that google does not care about users now. Or you should ask them what's exactly wrong. The current postmaster.google.com stats for alsa-project.org: User reported spam rate - 0% DKIM, DMARC, SFP success rate - all 100% Encryption - 100% IP reputation - bad Delivery errors - 40% - 100% Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

Re: [PATCH v3 00/10] Auxiliary bus implementation and SOF multi-client support
by Greg KH 23 Oct '20

23 Oct '20

On Fri, Oct 23, 2020 at 09:49:46AM +0300, Leon Romanovsky wrote: > On Thu, Oct 22, 2020 at 05:33:28PM -0700, Dave Ertman wrote: > > <...> > > > Dave Ertman (1): > > Add auxiliary bus support > > We are in merge window now and both netdev and RDMA are closed for > submissions. So I'll send my mlx5 conversion patches once -rc1 will > be tagged. > > However, It is important that this "auxiliary bus" patch will be applied > to some topic branch based on Linus's -rcX. It will give us an ability > to pull this patch to RDMA, VDPA and netdev subsystems at the same time. I will do that, but as you said, it's the middle of the merge window and I can't do anything until after -rc1 is out. Give us some time to catch up after that... greg k-h

1 0

on-board sound on ASUS TRX-40
by Dan Aloni 22 Oct '20

22 Oct '20

Hi, The on-board earphone jack does not seem to work on an ASUS TRX-40 board. Here's the alsa-info.sh output: http://alsa-project.org/db/?f=7a94c1b1eec4b2e623c75770364ec43c33d6c95c Tried coding up the patch below, but it _does not_ fix the problem. It does shows the earphone as 'plugged' though. Verified that it's not an hardware issue via Windows. Please instruct on how to debug this further. Thanks! diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9af7aa93f6fa..ab0f036f385e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -568,6 +568,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* ASUS TRX40-PRO */ + .id = USB_ID(0x0b05, 0x1918), + .map = trx40_mobo_map, + }, { /* Asrock TRX40 Creator */ .id = USB_ID(0x26ce, 0x0a01), .map = trx40_mobo_map, -- Dan Aloni

2 10

src/dlmisc.c:snd_dlpath_unlock() doesn't build if DL_ORIGIN_AVAILABLE is not defined
by GitHub issues - opened 22 Oct '20

22 Oct '20

alsa-project/alsa-lib issue #91 was opened from ollieparanoid: Hey all, thanks for maintaining alsa. In musl libc based Alpine Linux, the build of v1.2.4 fails with: ``` dlmisc.c:95:7: error: 'snd_plugin_dir_set' undeclared (first use in this function) 95 | if (!snd_plugin_dir_set) { | ^~~~~~~~~~~~~~~~~~ dlmisc.c:95:7: note: each undeclared identifier is reported only once for each function it appears in dlmisc.c:98:4: error: 'snd_plugin_dir' undeclared (first use in this function) 98 | snd_plugin_dir = strdup(env); | ``` This is a regression from https://github.com/alsa-project/alsa-lib/commit/8580c081c25678d11278efcb61b…, where the #ifdef...#else...#endif was removed. Issue URL : https://github.com/alsa-project/alsa-lib/issues/91 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Context expectations in ALSA
by Maxime Ripard 22 Oct '20

22 Oct '20

Hi! We currently have an issue reported by lockdep on the RaspberryPi and its HDMI audio output where, at startup, we end up scheduling in atomic context. This is caused by the HDMI driver polling some status bit that reports that the infoframes have been properly sent, and calling usleep_range between each iteration[1], and that is done in our trigger callback that seems to be run with a spinlock taken and the interrupt disabled (snd_pcm_action_lock_irq) as part of snd_pcm_start_lock_irq. This is the entire stack trace: # aplay --channels=2 -D 'iec958:CARD=vc4hdmi0,DEV=0' /root/test-tone.wav Playing WAVE '/root/test-tone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ 14.732730] BUG: sleeping function called from invalid context at drivers/gpu/drm/vc4/vc4_hdmi.c:276 [ 14.742005] in_atomic(): 1, irqs_disabled(): 128, non_block: 0, pid: 140, name: aplay [ 14.749955] CPU: 0 PID: 140 Comm: aplay Not tainted 5.9.0-rc5-v7l+ #66 [ 14.756578] Hardware name: BCM2711 [ 14.760026] Backtrace: [ 14.762524] [<c020d250>] (dump_backtrace) from [<c020d5f4>] (show_stack+0x20/0x24) [ 14.770209] r7:c167c9e4 r6:600e0093 r5:00000000 r4:c167c9e4 [ 14.775960] [<c020d5d4>] (show_stack) from [<c0703164>] (dump_stack+0xb8/0xd8) [ 14.783297] [<c07030ac>] (dump_stack) from [<c025108c>] (___might_sleep+0x138/0x17c) [ 14.791159] r10:00000084 r9:f0979b00 r8:c10ffd50 r7:00000010 r6:00000000 r5:00000114 [ 14.799105] r4:c10fe000 r3:600e0093 [ 14.802736] [<c0250f54>] (___might_sleep) from [<c0251140>] (__might_sleep+0x70/0xb0) [ 14.810680] r4:c0e8f5f4 [ 14.813256] [<c02510d0>] (__might_sleep) from [<c0840724>] (vc4_hdmi_stop_packet+0x120/0x330) [ 14.821907] r6:ef21f280 r5:00000003 r4:73e746f9 [ 14.826598] [<c0840604>] (vc4_hdmi_stop_packet) from [<c084422c>] (vc4_hdmi_write_infoframe+0x140/0x5d0) [ 14.836220] r9:f0979b00 r8:c10ffd50 r7:ef219900 r6:ef248840 r5:00000000 r4:ef21f280 [ 14.844084] [<c08440ec>] (vc4_hdmi_write_infoframe) from [<c0844718>] (vc4_hdmi_set_audio_infoframe+0x5c/0x80) [ 14.854236] r10:efac7738 r9:ef3ea240 r8:00000001 r7:ef219900 r6:ef248840 r5:ef21f040 [ 14.862180] r4:ef21f280 [ 14.864756] [<c08446bc>] (vc4_hdmi_set_audio_infoframe) from [<c0844774>] (vc4_hdmi_audio_trigger+0x38/0x238) [ 14.874815] r4:00000001 [ 14.877392] [<c084473c>] (vc4_hdmi_audio_trigger) from [<c0a155f0>] (snd_soc_pcm_dai_trigger+0x64/0xcc) [ 14.886923] r5:efac5a00 r4:00000000 [ 14.890555] [<c0a1558c>] (snd_soc_pcm_dai_trigger) from [<c0a18c70>] (soc_pcm_trigger+0x70/0xac) [ 14.899472] r9:ef3ea240 r8:c1133540 r7:00000003 r6:ef219900 r5:ef219900 r4:00000001 [ 14.907336] [<c0a18c00>] (soc_pcm_trigger) from [<c09fbd84>] (snd_pcm_do_start+0x3c/0x40) [ 14.915633] r5:c0c749dc r4:00000000 [ 14.919264] [<c09fbd48>] (snd_pcm_do_start) from [<c09fa390>] (snd_pcm_action_single+0x48/0x88) [ 14.928094] [<c09fa348>] (snd_pcm_action_single) from [<c09fc830>] (snd_pcm_action+0x6c/0x78) [ 14.936746] r7:00000003 r6:c0c749dc r5:00000000 r4:ef219900 [ 14.942493] [<c09fc7c4>] (snd_pcm_action) from [<c09fd7ec>] (snd_pcm_action_lock_irq+0x38/0x50) [ 14.951320] r7:00000030 r6:00000003 r5:c0c749dc r4:ef219900 [ 14.957069] [<c09fd7b4>] (snd_pcm_action_lock_irq) from [<c09ffd2c>] (snd_pcm_ioctl+0x930/0x14e4) [ 14.966073] r7:00000030 r6:00000000 r5:0085ee60 r4:ef219900 [ 14.971823] [<c09ff3fc>] (snd_pcm_ioctl) from [<c043accc>] (sys_ioctl+0xe4/0x9e4) [ 14.979420] r10:efac7738 r9:00000004 r8:c1133540 r7:0085ee60 r6:00005624 r5:c1133540 [ 14.987364] r4:00004142 [ 14.989940] [<c043abe8>] (sys_ioctl) from [<c0200040>] (ret_fast_syscall+0x0/0x28) [ 14.997621] Exception stack(0xc10fffa8 to 0xc10ffff0) [ 15.002749] ffa0: 00869260 b6fa8000 00000004 00004142 0085ee60 00000001 [ 15.011050] ffc0: 00869260 b6fa8000 00005624 00000036 00000000 00000000 00001589 00001589 [ 15.019350] ffe0: b6fa8504 bedb59fc b6f2ff68 b6da515c [ 15.024479] r10:00000036 r9:c10fe000 r8:c0200204 r7:00000036 r6:00005624 r5:b6fa8000 [ 15.032423] r4:00869260 We could switch to a udelay instead, but the idea of busy-waiting for up to a 100ms while having the interrupt disabled doesn't sound ideal either. It looks like the snd_soc_dai_link structure has a nonatomic flag that seems to be made to address more or less that issue, taking a mutex instead of a spinlock. However setting that flag results in another lockdep issue, since the dmaengine controller doing the DMA transfer would call snd_pcm_period_elapsed on completion, in a tasklet, this time taking a mutex in an atomic context which is just as bad as the initial issue. This is the stacktrace this time: # aplay --channels=2 -D 'iec958:CARD=vc4hdmi0,DEV=0' /root/test-tone.wav Playing WAVE '/root/test-tone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ 43.078900] BUG: sleeping function called from invalid context at kernel/locking/mutex.c:281 [ 43.087485] in_atomic(): 1, irqs_disabled(): 0, non_block: 0, pid: 0, name: swapper/0 [ 43.095452] CPU: 0 PID: 0 Comm: swapper/0 Not tainted 5.9.0-rc5-v7l+ #67 [ 43.102252] Hardware name: BCM2711 [ 43.105701] Backtrace: [ 43.108199] [<c020d250>] (dump_backtrace) from [<c020d5f4>] (show_stack+0x20/0x24) [ 43.115884] r7:c167c9e4 r6:600e0113 r5:00000000 r4:c167c9e4 [ 43.121636] [<c020d5d4>] (show_stack) from [<c0703164>] (dump_stack+0xb8/0xd8) [ 43.128972] [<c07030ac>] (dump_stack) from [<c025108c>] (___might_sleep+0x138/0x17c) [ 43.136833] r10:c1600000 r9:00000006 r8:00000000 r7:c1601d70 r6:00000000 r5:00000119 [ 43.144777] r4:c1600000 r3:600e0113 [ 43.148409] [<c0250f54>] (___might_sleep) from [<c0251140>] (__might_sleep+0x70/0xb0) [ 43.156353] r4:c0e32994 [ 43.158930] [<c02510d0>] (__might_sleep) from [<c0baf508>] (mutex_lock+0x2c/0x74) [ 43.166524] r6:efbfaa84 r5:ef26f548 r4:ef219998 [ 43.171216] [<c0baf4dc>] (mutex_lock) from [<c09fa82c>] (_snd_pcm_stream_lock_irqsave+0x2c/0x40) [ 43.180130] r5:ef26f548 r4:ef219900 [ 43.183765] [<c09fa800>] (_snd_pcm_stream_lock_irqsave) from [<c0a03848>] (snd_pcm_period_elapsed+0x20/0xa4) [ 43.193743] [<c0a03828>] (snd_pcm_period_elapsed) from [<c0a067d4>] (dmaengine_pcm_dma_complete+0x54/0x58) [ 43.203539] r7:c1601d70 r6:efbfaa84 r5:ef26f548 r4:00000000 [ 43.209290] [<c0a06780>] (dmaengine_pcm_dma_complete) from [<c07885e8>] (vchan_complete+0x238/0x240) [ 43.218567] [<c07883b0>] (vchan_complete) from [<c0228f9c>] (tasklet_action_common.constprop.0+0x84/0x12c) [ 43.228365] r10:c1600000 r9:00000006 r8:00000000 r7:dac802e0 r6:efbfaa84 r5:00000000 [ 43.236309] r4:efbfaa80 [ 43.238886] [<c0228f18>] (tasklet_action_common.constprop.0) from [<c022906c>] (tasklet_action+0x28/0x30) [ 43.248596] r9:00000040 r8:ef80b800 r7:00000101 r6:00000006 r5:00000007 r4:c1603098 [ 43.256459] [<c0229044>] (tasklet_action) from [<c0201284>] (__do_softirq+0x18c/0x47c) [ 43.264498] [<c02010f8>] (__do_softirq) from [<c0228de4>] (irq_exit+0xd0/0xf8) [ 43.271829] r10:c1051f40 r9:c1600000 r8:ef80b800 r7:00000001 r6:00000000 r5:00000000 [ 43.279774] r4:ffffe000 [ 43.282349] [<c0228d14>] (irq_exit) from [<c0287c54>] (__handle_domain_irq+0x70/0xc0) [ 43.290294] r5:00000000 r4:c148ee80 [ 43.293927] [<c0287be4>] (__handle_domain_irq) from [<c071d8b4>] (gic_handle_irq+0x4c/0x88) [ 43.302403] r9:c1600000 r8:c1601ed0 r7:f0803000 r6:f0802000 r5:f080200c r4:c16058d8 [ 43.310266] [<c071d868>] (gic_handle_irq) from [<c0200abc>] (__irq_svc+0x5c/0x7c) [ 43.317856] Exception stack(0xc1601ed0 to 0xc1601f18) [ 43.322983] 1ec0: c0208b84 00000000 19bf8000 600e0093 [ 43.331283] 1ee0: c1600000 00000000 c1604e34 c1604e7c c16d20e5 c0e30cac c1051f40 c1601f2c [ 43.339582] 1f00: ffffe000 c1601f20 c1600000 c0208b88 600e0013 ffffffff [ 43.346297] r9:c1600000 r8:c16d20e5 r7:c1601f04 r6:ffffffff r5:600e0013 r4:c0208b88 [ 43.354164] [<c0208b54>] (arch_cpu_idle) from [<c0bb23c8>] (default_idle_call+0x3c/0x184) [ 43.362469] [<c0bb238c>] (default_idle_call) from [<c025b5f8>] (do_idle+0x110/0x170) [ 43.370327] r5:00000000 r4:c1600000 [ 43.373958] [<c025b4e8>] (do_idle) from [<c025b950>] (cpu_startup_entry+0x28/0x2c) [ 43.381641] r9:00000193 r8:effff900 r7:c1604e00 r6:00000000 r5:c160ec50 r4:000000d7 [ 43.389503] [<c025b928>] (cpu_startup_entry) from [<c0baa65c>] (rest_init+0x108/0x110) [ 43.397542] [<c0baa554>] (rest_init) from [<c1000cc8>] (arch_call_rest_init+0x18/0x1c) [ 43.405575] r5:c1051f40 r4:c16e7078 [ 43.409205] [<c1000cb0>] (arch_call_rest_init) from [<c1001538>] (start_kernel+0x7f4/0x834) [ 43.417682] [<c1000d44>] (start_kernel) from [<00000000>] (0x0) So, I'm not really sure what I'm supposed to do here. The drivers involved don't appear to be doing anything extraordinary, but the issues lockdep report are definitely valid too. What are the expectations in terms of context from ALSA when running the callbacks, and how can we fix it? Thanks! Maxime 1: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/dri…

4 7

[PATCH v2 0/5] ASoC: Mediatek: Add support for MT8192 SoC
by Jiaxin Yu 22 Oct '20

22 Oct '20

This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same time, the calibration function of MT6359 is completed with real machine driver. The patch is based on broonie tree "for-next" branch. Change since v1: - fixed some typos related to dt-bindings in [v1,3/5] and [v1,5/5] - add vendor prefix to the properties, such as "mediatek,apmixedsys" - add a dependency description to indicate the required header files Jiaxin Yu (5): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document .../bindings/sound/mt8192-afe-pcm.yaml | 103 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 42 + sound/soc/codecs/mt6359.c | 110 + sound/soc/codecs/mt6359.h | 7 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 ++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 + .../soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 306 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1489 ++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2139 +++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 ++++ .../mediatek/mt8192/mt8192-interconnection.h | 65 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 1058 ++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 +++++++++++++++++ 23 files changed, 13341 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h -- 2.18.0

3 11

[PATCH v11 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 22 Oct '20

22 Oct '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name SC7180_LPASS_DP defined in sc7180-lpass.h. It will be posted in the newer patchset of https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Changes from v9 to v10 - Documentation: - Let compatible string be more specific for board configuration to allow for future changes. - Machine driver: - Fixed unused include and macro. - Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h. - Let sound card be dynamically allocated. - Change compatible string accordingly. Changes from v10 to v11 - Machine driver: - Use temporary macro LPASS_DP_RX for future change in sc7180-lpass.h. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 266 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 421 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.618.gf4bc123cb7-goog

5 21

Fix muted output and low ouput/input volume on a Dell Latitude 3410
by GitHub pull_request - opened 22 Oct '20

22 Oct '20

alsa-project/alsa-ucm-conf pull request #56 was opened from J-Bu: Using following environment on a Dell Latitude 3410 results in muted master channel and low output/input volume: - alsa-libs 1.2.4 - alsa-ucm-conf 1.2.4 - alsa-topology-conf 1.2.4 - alsa-utils 1.2.4 - kernel 5.9.1 -> with commit 3dca35e applied on top of it - sof-bin v1.6 The propused changes fixes the problem for me. I don't know why but the given values are not 1:1 mapped to levels in alsamixer, please see attached iamges for alsamixer levels before and after the change, the alsamixer levels where captured after running "alsactl init": before: ![before](https://user-images.githubusercontent.com/15876551/96853514-bf0f5f80-145a-11eb-9c7f-751834a56491.png) after: ![after](https://user-images.githubusercontent.com/15876551/96853530-c46caa00-145a-11eb-9057-5399382d195d.png) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/56 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/56.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

How to downmix 5.1 and 7.1 channels to 2 channels
by Amanda Ava Koci Pereira 22 Oct '20

22 Oct '20

Hi guys... We need some clues about to downmix 5.1 and 7.1 channels to 2 channels for the moment we are using the following code -> It works but fails miserably if all channels have data at same time #define AR_K_P_ATTENUATION_LEVEL 1.0 int p = 0; int i; int save_channel_0_k_p; int save_channel_1_k_p; double save_channel_0_data_double_k_p; double save_channel_1_data_double_k_p; double sample_double_temp_k_p; int number_of_samples_k_p = decoded_frame->nb_samples; int ricardo_deslocador__ = 0; frame_processed_k_p = 1; int outputBufferLen = number_of_samples_k_p * 2 * 2; *size_out=outputBufferLen; short *outputBuffer = (int16_t *)buf; int16_t * ar_buffer_ = (int16_t *) decoded_frame->data[0]; for(i=0; i < number_of_samples_k_p; i++) { //channel 0 int16_t sample_k_p = ar_buffer_[ricardo_deslocador__++]; save_channel_0_k_p = p; save_channel_0_data_double_k_p = (double) sample_k_p; p++; /////////////////////////////////////////////// //channel 1 sample_k_p = ar_buffer_[ricardo_deslocador__++]; save_channel_1_k_p = p;//so usa no final... save_channel_1_data_double_k_p = (double) sample_k_p; p++; //////////////////////////////////////////////// //center sample_k_p = ar_buffer_[ricardo_deslocador__++]; sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_0_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = -32768.0; } sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_1_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = -32768.0; } ////////////////////////////////////////////////////// ///////subwoffer sample_k_p = ar_buffer_[ricardo_deslocador__++]; sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_0_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = -32768.0; } sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_1_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = -32768.0; } //////////////////////////////////////////////////// /////surround left 4 sample_k_p = ar_buffer_[ricardo_deslocador__++]; sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_0_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_0_data_double_k_p) { save_channel_0_data_double_k_p = -32768.0; } /////////////////////////////////////////////// //////////surround right 5 sample_k_p = ar_buffer_[ricardo_deslocador__++]; sample_double_temp_k_p = (double) sample_k_p; sample_double_temp_k_p *= AR_K_P_ATTENUATION_LEVEL; save_channel_1_data_double_k_p += sample_double_temp_k_p; if(32767.0 < save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = 32767.0; } if(-32768.0 > save_channel_1_data_double_k_p) { save_channel_1_data_double_k_p = -32768.0; } ///////////////////////////////////////////// //finalizing outputBuffer[save_channel_0_k_p] = (signed short) save_channel_0_data_double_k_p; outputBuffer[save_channel_1_k_p] = (signed short) save_channel_1_data_double_k_p; } /* if possible point us to the files in ALSA that make the downmix and it will be enough Thanks a lot */ Ricardo

2 2

[PATCH] ALSA: hda - Fix the return value if cb func is already registered
by Hui Wang 22 Oct '20

22 Oct '20

If the cb function is already registered, should return the pointer of the structure hda_jack_callback which contains this cb func, but instead it returns the NULL. Now fix it by replacing func_is_already_in_callback_list() with find_callback_from_list(). Fixes: f4794c6064a8 ("ALSA:hda - Don't register a cb func if it is registered already") Reported-and-suggested-by: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_jack.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index ded4813f8b54..588059428d8f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -275,16 +275,21 @@ int snd_hda_jack_detect_state_mst(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state_mst); -static bool func_is_already_in_callback_list(struct hda_jack_tbl *jack, - hda_jack_callback_fn func) +static struct hda_jack_callback * +find_callback_from_list(struct hda_jack_tbl *jack, + hda_jack_callback_fn func) { struct hda_jack_callback *cb; + if (!func) + return NULL; + for (cb = jack->callback; cb; cb = cb->next) { if (cb->func == func) - return true; + return cb; } - return false; + + return NULL; } /** @@ -309,7 +314,10 @@ snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid, jack = snd_hda_jack_tbl_new(codec, nid, dev_id); if (!jack) return ERR_PTR(-ENOMEM); - if (func && !func_is_already_in_callback_list(jack, func)) { + + callback = find_callback_from_list(jack, func); + + if (func && !callback) { callback = kzalloc(sizeof(*callback), GFP_KERNEL); if (!callback) return ERR_PTR(-ENOMEM); -- 2.25.1

2 1

PCM link state
by Matt Flax 22 Oct '20

22 Oct '20

Is there a way to check whether PCMs are linked ? Linking is done with snd_pcm_link : https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gac6c3309… But this is no function to tell if a pcm is linked. Matt

1 0

[bug report] ALSA: hda - Don't register a cb func if it is registered already
by Dan Carpenter 22 Oct '20

22 Oct '20

Hello Hui Wang, The patch f4794c6064a8: "ALSA: hda - Don't register a cb func if it is registered already" from Sep 30, 2020, leads to the following static checker warning: sound/pci/hda/patch_sigmatel.c:3075 stac92hd71bxx_fixup_hp_m4() warn: 'jack' can also be NULL sound/pci/hda/patch_sigmatel.c 3069 /* Enable VREF power saving on GPIO1 detect */ 3070 snd_hda_codec_write_cache(codec, codec->core.afg, 0, 3071 AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); 3072 jack = snd_hda_jack_detect_enable_callback(codec, codec->core.afg, 3073 stac_vref_event); Originally snd_hda_jack_detect_enable_callback() would not return NULL here. 3074 if (!IS_ERR(jack)) 3075 jack->private_data = 0x02; 3076 3077 spec->gpio_mask |= 0x02; But now we have this: sound/pci/hda/hda_jack.c 301 struct hda_jack_callback * 302 snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid, 303 int dev_id, hda_jack_callback_fn func) 304 { 305 struct hda_jack_tbl *jack; 306 struct hda_jack_callback *callback = NULL; 307 int err; 308 309 jack = snd_hda_jack_tbl_new(codec, nid, dev_id); 310 if (!jack) 311 return ERR_PTR(-ENOMEM); 312 if (func && !func_is_already_in_callback_list(jack, func)) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ We only allocate callback if there isn't one already. 313 callback = kzalloc(sizeof(*callback), GFP_KERNEL); 314 if (!callback) 315 return ERR_PTR(-ENOMEM); 316 callback->func = func; 317 callback->nid = jack->nid; 318 callback->dev_id = jack->dev_id; 319 callback->next = jack->callback; 320 jack->callback = callback; 321 } 322 323 if (jack->jack_detect) 324 return callback; /* already registered */ ^^^^^^^^^^^^^^^ So presumably this should be jack->callback 325 jack->jack_detect = 1; 326 if (codec->jackpoll_interval > 0) 327 return callback; /* No unsol if we're polling instead */ ^^^^^^^^^^^^^^^^ 328 err = snd_hda_codec_write_cache(codec, nid, 0, 329 AC_VERB_SET_UNSOLICITED_ENABLE, 330 AC_USRSP_EN | jack->tag); 331 if (err < 0) 332 return ERR_PTR(err); 333 return callback; ^^^^^^^^^^^^^^^^ And these as well. 334 } regards, dan carpenter

3 6

[PATCH v3 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 22 Oct '20

22 Oct '20

Hi Rob, Mark Cc: Sameer These are v3 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 165 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 623 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

1 1

[GIT PULL] ASoC updates for v5.10
by Mark Brown 21 Oct '20

21 Oct '20

The following changes since commit 549738f15da0e5a00275977623be199fbbf7df50: Linux 5.9-rc8 (2020-10-04 16:04:34 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.10 for you to fetch changes up to c890e30b069a2792a5a34e8510a7a437dd6f5b3d: Merge remote-tracking branch 'asoc/for-5.10' into asoc-next (2020-10-09 15:42:31 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.10 Not a huge amount going on in the core for ASoC this time but quite a lot of driver activity, especially for the Intel platforms: - Replacement of the DSP driver for some older x86 systems with a new one which was written with closer reference to the DSP firmware so should hopefully be more robust and maintainable. - A big batch of static checker and other fixes for the rest of the x86 DSP drivers. - Cleanup of the error unwinding code from Morimoto-san, hopefully making it more robust. - Helpers for parsing auxiluary devices from the device tree from Stephan Gerhold. - New support for AllWinner A64, Cirrus Logic CS4234, Mediatek MT6359 Microchip S/PDIF TX and RX controllers, Realtek RT1015P, and Texas Instruments J721E, TAS2110, TAS2564 and TAS2764 ---------------------------------------------------------------- Adam Brickman (1): ASoC: wm_adsp: Pass full name to snd_ctl_notify Adam Thomson (4): ASoC: da7219: Move required devm_* allocations to device level code ASoC: da7219: Move soft reset handling to codec level probe ASoC: da7219: Fix clock handling around codec level probe ASoC: da7219: Fix I/O voltage range configuration during probe Ajit Pandey (4): ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-platform: Replace card->dev with component->dev ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio ASoC: Add sc7180-lpass bindings header Akshu Agrawal (1): ASoC: AMD: Clean kernel log from deferred probe error messages Alex Dewar (1): ASoC: omap-mcbsp: Fix use of uninitialised pointer Alexandre Belloni (2): spi: atmel: remove unnecessary include ASoC: atmel-pcm: remove unnecessary include Anson Huang (1): ASoC: Convert NXP spdif to json-schema Arnd Bergmann (4): ASoC: samsung: h1940: turn into platform driver ASoC: samsung: neo1973: turn into platform driver ASoC: samsung: rx1950: turn into platform driver ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS Bard Liao (7): ASoC: intel: sof_sdw: add .exit callback function ASoC: Intel: modify SoundWire version id in acpi match table ASoC: Intel: sof_sdw: check SoundWire version when matching codec ASoC: Intel: sof_sdw: rename id as part_id ASoC: intel: sof_sdw: add rt711 rt1316 rt714 SDCA codec support. ASoC: Intel: sof_sdw: clean-up inclusion of header files ASoC: Intel: add support for new SoundWire hardware layout on TGL Brent Lu (2): ASoC: hdac_hdmi: support 'ELD' mixer ASoC: hdac_hdmi: remove cancel_work_sync in runtime suspend Camel Guo (3): ASoC: tlv320adcx140: Fix digital gain range dt-bindings: tlv320adcx140: Add GPIO config and drive config ASoC: tlv320adcx140: Add support for configuring GPIO pin Cezary Rojewski (34): spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers ASoC: Intel: Add catpt base members ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC message handlers ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Device driver lifecycle ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: bdw-rt5650: Mark FE DAIs as nonatomic ASoC: Intel: bdw-rt5677: Mark FE DAIs as nonatomic ASoC: Intel: broadwell: Mark FE DAIs as nonatomic ASoC: Intel: haswell: Mark FE DAIs as nonatomic ASoC: Intel: Remove haswell solution ASoC: Intel: Remove max98090 support for baytrail solution ASoC: Intel: Remove rt5640 support for baytrail solution ASoC: Intel: Remove baytrail solution ASoC: Intel: Remove SST ACPI component ASoC: Intel: Remove SST firmware components ASoC: Intel: Skylake: Unassign ram_read and read_write ops ASoC: Intel: Remove unused DSP operations ASoC: Intel: Remove unused DSP interface fields ASoC: Intel: Remove SST-legacy specific constants ASoC: Intel: Make atom components independent of sst-dsp ASoC: Intel: Remove sst_pdata structure ASoC: Intel: Remove sst_dsp_get_thread_context ASoC: Intel: catpt: Fix compilation when CONFIG_MODULES is disabled ASoC: Intel: catpt: Add explicit DMADEVICES kconfig dependency Charles Keepax (2): ASoC: wm_adsp: Update naming in error handling ASoC: cs4234: Add dtschema binding document Cheng-Yi Chiang (1): ASoC: hdmi-codec: Use set_jack ops to set jack Christophe JAILLET (1): ASoC: wm8523: Fix a typo in a comment Codrin Ciubotariu (6): ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller ASoC: add DT bindings for Microchip S/PDIF TX Controller ASoC: mchp-spdifrx: add driver for SPDIF RX ASoC: add DT bindings for Microchip S/PDIF RX Controller ASoC: mchp-spdifrx: convert to devm_platform_get_and_ioremap_resource ASoC: mchp-spdiftx: remove 'TX' from playback stream name Colin Ian King (3): sound: remove duplicate "the the" phrase in Kconfig text ASoC: qcom: add missing out of memory check on drvdata->clks allocation ASoC: mchp-spdifrx: fix spelling mistake "overrrun" -> "overrun" Dan Carpenter (1): ASoC: hdac_hdmi: tidy up a memset() Dan Murphy (29): dt-bindings: tas2562: Remove tas2562 text file dt-bindings: tas2562: Add device specification links ASoC: tlv320adcx140: Add digital mic channel enable routing ASoC: tlv320adcx140: Idle the device while writing registers ASoC: tlv320adcx140: Fix BCLK inversion for DSP modes ASoC: tlv320adcx140: Add the config to configure Tx ASI output ASoC: tlv320adcx140: Add ASI Tx drive dt-bindings: tas2562: Add TAS2564 to binding ASoC: tas2562: Add the TAS2564 compatible ASoC: tas2770: Fix calling reset in probe ASoC: tas2770: Add missing bias level power states ASoC: tas2770: Fix required DT properties in the code ASoC: tas2770: Fix error handling with update_bits dt-bindings: tas2562: Add the TAS2110 amplifier ASoC: tas2562: Add the TAS2110 class-D amplifier dt-bindings: tas2770: Fix I2C addresses for the TAS2770 ASoC: tas2770: Fix unbalanced calls to pm_runtime ASoC: tas2770: Convert bit mask to GENMASK in header ASoC: tas2770: Fix the spacing and new lines ASoC: tas2770: Refactor sample rate function ASoC: tas2562: Remove duplicate code for I/V sense dt-bindings: tas2770: Add shutdown gpio property ASoC: tas2770: Add shutdown capability via a GPIO ASoC: tas2770: Set regcache when shutting down and waking device ASoC: tas2770: Remove ti,asi-format code ASoC: tas2770: Remove unused variables dt-bindings: tas2770: Mark ti,asi-format to deprecated dt-bindings: tas2764: Add the TAS2764 binding doc ASoC: tas2764: Add the driver for the TAS2764 Daniel Baluta (3): ASoC: SOF: pm: Fix prepare callback behavior for OF usecase ASoC: SOF: Activate runtime PM with SOF OF device ASoC: SOF: Add .prepare/.complete callbacks Dharageswari R (2): ASoC: topology: Add support for WO and RO TLV byte kcontrols ASoC: SOF: Implement snd_sof_bytes_ext_volatile_get kcontrol IO Dmitry Baryshkov (1): ASoC: core: remove artificial component and DAI name constraint Douglas Anderson (2): ASoC: rt5682: Prefer async probe ASoC: qcom: lpass-sc7180: Add MODULE_DEVICE_TABLE Guennadi Liakhovetski (7): ASoC: SOF: topology: (cosmetic) remove redundant variable initialisations ASoC: SOF: (cosmetic) use the "bool" type where it makes sense ASoC: SOF: add a "core" parameter to widget loading functions ASoC: SOF: support topology components on secondary cores ASoC: SOF: (cosmetic) remove redundant "ret" variable uses ASoC: SOF: remove several superfluous type-casts ASoC: SOF: fix range checks Guillermo Rodríguez (1): ASoC: cs42l51: add additional ADC volume controls Iulian Olaru (4): ASoC: SOF: loader: Add debug box region ASoC: SOF: imx: Replace sdev->private with sdev->pdata->hw_pdata ASoC: SOF: sof-of-dev: Add .arch_ops field ASoC: SOF: imx: Add debug support for imx platforms Jack Yu (1): ASoC: rt715: Add power-up delay to fix dmic pop sound issue. Jaska Uimonen (1): ASoC: SOF: intel: hda: support also devices with 1 and 3 dmics Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document Johan Jonker (1): ASoC: rockchip-spdif: add description for rk3308 Kai Vehmanen (2): ASoC: SOF: imx: add missing MODULE_LICENSE() for imx-common ASoC: hdac: make SOF HDA codec driver probe deterministic Karol Trzcinski (3): ASoC: SOF: IPC: make sof_ipc_window monosized ASoC: SOF: ext_manifest: Parse debug ABI version ASoC: SOF: Add `src_hash` to `sof_ipc_fw_version` structure Keyon Jie (18): ASoC: SOF: topology: fix the ipc_size calculation for process component ASoC: SOF: tokens: add token for component UUID ASoC: SOF: add comp_ext to struct snd_sof_widget ASoC: SOF: topology: create component extended tokens ASoC: SOF: topology: parse comp_ext_tokens for all widgets ASoC: SOF: use the sof_ipc_comp reserved bytes for extended data ASoC: SOF: topology: add helper for setting up IPC component ASoC: SOF: append extended data to sof_ipc_comp_dai ASoC: SOF: append extended data to sof_ipc_comp_mixer ASoC: SOF: append extended data to sof_ipc_comp_volume ASoC: SOF: append extended data to sof_ipc_comp_host ASoC: SOF: append extended data to sof_ipc_comp_src ASoC: SOF: append extended data to sof_ipc_comp_asrc ASoC: SOF: append extended data to sof_ipc_comp_tone ASoC: SOF: append extended data to sof_ipc_comp_process ASoC: SOF: append extended data to sof_ipc_comp_mux ASoC: SOF: topology: make process type optional ASoC: SOF: topology: fix the process being scheduled on core0 always Krzysztof Kozlowski (8): ASoC: samsung-i2s: Use unevaluatedProperties ASoC: midas-audio: Correct parsing sound-dai phandles ASoC: odroid: Use unevaluatedProperties ASoC: odroid: Add missing properties ASoC: samsung-i2s: Add missing properties ASoC: dt-bindings: aries-wm8994: Match compatibles with enum ASoC: dt-bindings: Correct interrupt flags in examples ASoC: dt-bindings: sgtl5000: Add common clock properties Kuninori Morimoto (10): ASoC: soc-core: add snd_soc_find_dai_with_mutex() ASoC: soc-dai: add mark for snd_soc_dai_startup/shutdown() ASoC: soc-link: add mark for snd_soc_link_startup/shutdown() ASoC: soc-component: add mark for soc_pcm_components_open/close() ASoC: soc-component: add mark for snd_soc_pcm_component_pm_runtime_get/put() ASoC: soc-pcm: add soc_pcm_clean() and call it from soc_pcm_open/close() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_dai_startup() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_component_module/open() ASoC: soc-pcm: ignore un-needed mutex_unlock() case on soc_pcm_open() ASoC: soc-pcm: add missing ret=0 at soc_pcm_open() Lucas Tanure (1): ASoC: cs4234: Add support for Cirrus Logic CS4234 codec Mark Brown (52): Merge existing fixes from asoc/for-5.9 Merge series "ASoC: SOF/Intel: fix cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt5682: Use clk APIs better" from Stephen Boyd <swboyd(a)chromium.org>: Merge series "Allwinner A64 digital audio codec fixes" from Samuel Holland <samuel(a)sholland.org>: Merge series "This patch series is to enable multiple features on the Keembay Platform" from Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com>: Merge series "ASoC: sun50i-codec-analog: Cleanup and power management" from Samuel Holland <samuel(a)sholland.org>: Merge series "Codec workaround" from Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com>: Merge series "ASoC: qcom: Add support for SC7180 lpass variant" from Rohit kumar <rohitkr(a)codeaurora.org>: Merge series "refine and clean code for synchronous mode" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "ASoC: da7219: Reorganise device/codec level probe/remove" from Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com>: Merge series "ASoC: Intel: fix cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: qdsp6: add gapless compressed audio support" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge tag 'samsung-platdrv-boards' into asoc-5.10 Merge series "Add mediatek codec mt6359 driver" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: Intel: machine driver updates for 5.10" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge tag 'v5.9-rc2' into asoc-5.10 Merge series "ASoC: SOF: trivial code/log/comment improvements" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "SOF fixes and updates" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "SOF fixes and updates for FW boot" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: SOF: multi core support for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: Add sdw stream operations to dailink ops." from Bard Liao <yung-chuan.liao(a)linux.intel.com>: Merge series "ASoC: SOF: component UUID support for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: sun8i-codec driver cleanup" from Samuel Holland <samuel(a)sholland.org>: Merge series "ASoC: qcom: common: Parse auxiliary devices from device tree" from Stephan Gerhold <stephan(a)gerhold.net>: Merge branch 'topic/tasklet-convert' of https://git.kernel.org/.../tiwai/sound into asoc-5.10 Merge series "ASoC: q6dsp: Add support to Codec Ports." from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: ti: j721e-evm: Support for j7200 variant" from Peter Ujfalusi <peter.ujfalusi(a)ti.com>: Merge series "ASoC: mediatek: mt8183-da7219: support machine driver for rt1015p" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: SOF: DSP core management fixes for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge branch 'asoc-5.9' into asoc-5.10 Merge series "ASoC: q6afe: add clocks support" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: SOF: small fixes for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: fsl_sai: update the register list" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge branch 'asoc-5.9' into asoc-5.10 Merge series "ASoC: SOF: fix kcontrol size checks" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: Intel: sdw machine driver updates for 5.10" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "ASoC: qdsp6: fix some warnings when build without CONFIG_OF" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "Enable runtime PM for SOF device" from Daniel Baluta <daniel.baluta(a)oss.nxp.com> Daniel Baluta <daniel.baluta(a)nxp.com>: Merge tag 'regmap-field-bulk-api' of https://git.kernel.org/.../broonie/regmap into asoc-5.10 Merge series "ASoC: merge soc_pcm_open() rollback and soc_pcm_close()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Intel: Catpt - Lynx and Wildcat point" from Cezary Rojewski <cezary.rojewski(a)intel.com>: ASoC: fsl_spdif: Remove unused np Merge series "Add driver for Microchip S/PDIF RX" from Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com>: Merge series "ASoC: Intel: sof_sdw: minor corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: sun8i-codec: support for AIF2 and AIF3" from Samuel Holland <samuel(a)sholland.org>: Merge series "ASoC: Intel: Remove obsolete solutions and components" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge tag 'v5.9-rc5' into asoc-5.10 Merge series "Qualcomm's lpass-hdmi ASoC driver to support audio over dp port" from Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org>: Merge series "ASoC: stm32: dfsdm: change rate limits" from Olivier Moysan <olivier.moysan(a)st.com>: ASoC: dmaengine: Document support for TX only or RX only streams Merge remote-tracking branch 'asoc/for-5.9' into asoc-linus Merge remote-tracking branch 'asoc/for-5.10' into asoc-next Matthias Schiffer (4): ASoC: bindings: fsl-asoc-card: add compatible string for TLV320AIC32x4 codec ASoC: fsl-asoc-card: add support for TLV320AIC32x4 codec ASoC: codec: tlv320aic32x4: fix missing aic32x4_disable_regulators() in error path ASoC: codec: tlv320aic32x4: do software reset before clock registration Michael Sit Wei Hong (6): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel, keembay-i2s: Add new compatible string ASoC: Intel: KeemBay: Fix warning potential ! vs ~ typo Miquel Raynal (3): ASoC: tlv320aic32x4: Ensure a minimum delay before clock stabilization ASoC: tlv320aic32x4: Fix bdiv clock rate derivation ASoC: tlv320aic32x4: Enable fast charge Necip Fazil Yildiran (1): ASoC: cros_ec_codec: fix kconfig dependency warning for SND_SOC_CROS_EC_CODEC Oder Chiou (1): ASoC: rt5682: Enable the power of "MICBIAS" and "Vref2" for the DMIC clock Olivier Moysan (3): ASoC: stm32: sai: add pm_runtime support ASoC: stm32: dfsdm: change rate limits ASoC: stm32: dfsdm: add actual resolution trace Pavel Dobias (1): ASoC: max9867: shutdown codec when changing filter type Peter Ujfalusi (5): ASoC: ti: Kconfig: Allow the j721e machine driver to be used on K3 platform ASoC: dt-bindings: ti, j721e-cpb-audio: Document support for j7200-cpb ASoC: ti: j721e-evm: Add support for j7200-cpb audio ASoC: ti: j721e-evm: Fix compiler warning when CONFIG_OF=n ASoC: ti: davinci-mcasp: Use &pdev->dev for early dev_warn Pierre-Louis Bossart (59): ASOC: SOF: Intel: hda-codec: move unused label to correct position ASoC: SOF: Intel: hda-codec: move variable used conditionally ASoC: Intel: rename shadowed variable for all broadwell boards ASoC: Intel: bytcht_cx2072x: simplify return handling ASoC: Intel: Atom: sst-atom-controls: remove redundant assignments ASoC: Intel: Atom: compress: remove redundant assignment ASoC: Intel: Atom: platform-pcm: remove redundant assignment ASoC: Intel: Atom: sst: remove useless NULL assignment ASoC: Intel: Atom: remove redundant initialization ASoC: Intel: Atom: sst_pvt: remove redundant initialization ASoC: Intel: Atom: platform-pcm: fix redundant return ASoC: Intel: Atom: remove useless assignment ASoC: Intel: Atom: sst_loader: remove always-true condition ASoC: Intel: Atom: sst_pvt: simplify return handling ASoC: Intel: Atom: (cosmetic) align parameters ASoC: Intel: Baytrail: (cosmetic) align function parameters ASoC: Intel: common: (cosmetic) align function parameters ASoC: Intel: haswell: (cosmetic) align function parameters ASoC: Intel: haswell-ipc: remove redundant assignments ASoC: Intel: Skylake: skl-nhlt: remove redundant initialization ASoC: Intel: Skylake: cldma: remove redundant initialization ASoC: Intel: Skylake: sst-utils: remove redundant assignment ASoC: Intel: Skylake: skl-topology: remove redundant assignments ASoC: Intel: Skylake: skl-topology: remove redundant assignment ASoC: Intel: Skylake: (cosmetic) align function parameters ASoC: codecs: wm0010: use DECLARE_COMPLETION_ONSTACK() macro ASoC: codecs: rt*-sdw: use SDW_SLAVE_ENTRY_EXT ASoC: SOF: Intel: hda: import SOUNDWIRE_INIT namespace ASoC: SOF: Intel: add build support for SoundWire ASoC: Intel: soc-acpi: cnl: add support for rt5682 on SoundWire link2 ASoC: Intel: sof-soundwire: add support for rt5682 on link2 ASoC: Intel: soc-acpi: mirror CML and TGL configurations ASoC: Intel: soc-acpi: add support for SDCA boards ASoC: codecs: max98373-sdw: add missing test on resume ASoC: SOF: IPC: reduce verbosity of IPC pointer updates ASoC: SOF: acpi: add dev_dbg() log for probe completion ASoC: SOF: Intel: add dev_dbg log when driver is not selected ASoC: Intel: use consistent HDAudio spelling in comments/docs ASoC: SOF: loader: fix memory leak in get_ext_windows ASoC: soc-dai: clarify return value for get_sdw_stream() ASoC: Intel: sof_sdw: add dailink .trigger callback ASoC: Intel: sof_sdw: add dailink .prepare and .hw_free callback ASoC: SOF: Intel: hda-loader: s/master/primary ASoC: topology: disable size checks for bytes_ext controls if needed ASoC: SOF: debug: update test for pm_runtime_get_sync() ASoC: SOF: control: update test for pm_runtime_get_sync() ASoC: SOF: Intel: hda: reduce verbosity of boot error logs ASoC: SOF: control: fix size checks for ext_bytes control .get() ASoC: SOF: control: fix size checks for volatile ext_bytes control .get() ASoC: SOF: control: add size checks for ext_bytes control .put() ASoC: Intel: sof_sdw: remove ternary operator ASoC: Intel: add codec name prefix to ACPI machine description ASoC: Intel: sof_sdw: remove hard-coded codec_conf table ASoC: Intel: sof_sdw_rt700: add codec prefix ASoC: Intel: hda_dsp_common: use static function in conditional block ASoC: Intel: sof_sdw_rt1308: add extra check on init ASoC: Intel: sof_sdw_rt1316: add missing component string ASoC: rt715-sdw: probe with RT714 Device ID ASoC: Intel: sof_sdw: add version_id to avoid rt714/rt715 confusion Qinglang Miao (7): ASoC: soc-core: use devm_snd_soc_register_card() ASoC: fsl: imx-mc13783: use devm_snd_soc_register_card() ASoC: tegra: tegra_max98090: use devm_snd_soc_register_card() ASoC: tegra: tegra_wm8753: use devm_snd_soc_register_card() ASoC: tegra: tegra_rt5640: use devm_snd_soc_register_card() ASoC: fsl: mx27vis-aic32x4: use devm_snd_soc_register_card() ASoC: tegra: trimslice.c: use devm_snd_soc_register_card() Rander Wang (5): ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case SoC: Intel: sof_sdw: Add support for product Ripto ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work ASoC: SOF: Intel: Add support for tgl-h ASOC: Intel: sof_sdw: restore playback functionality with max98373 amps Randy Dunlap (4): ASoC: various vendors: delete repeated words in comments ASoC: codecs: delete repeated words in comments ASoC: SOF: delete repeated words in comments ASoC: qcom: fix SDM845 & QDSP6 dependencies more Ranjani Sridharan (15): ASoC: SOF: topology: remove unnecessary memory alloc for sdev->private ASoC: SOF: topology: reduce the log level for unhandled widgets ASoC: SOF: Intel: hda: report error only for the last ROM init iteration ASoC: SOF: Intel: hda: add extended rom status dump to error log ALSA: hda: fix VS_LTRC register name ASoC: SOF: Intel: hda: Add helper function to program ICCMAX stream ASoC: SOF: Intel: hda: modify the signature of get_stream_with_tag() ASoC: SOF: Intel: hda: define macro for code loader stream format ASoC: SOF: Intel: hda: Define FW boot sequence with ICCMAX ASoC: SOF: Intel: hda: Add sof_tgl_ops for TGL platforms ASoC: SOF: Intel: hda: Simplify error handling during FW boot ASoC: SOF: topology: fix core enable sequence ASoC: SOF: rename cores_mask to host_managed_cores_mask ASoC: SOF: Intel: hda: modify core_power_up/down op ASoC: SOF: Intel: remove the HDA_DSP_CORE_MASK() macro Richard Fitzgerald (3): ASoC: cs47l15: Fix EPOUT->HPOUT1 Mono Mux routing ASoC: cs47l35: Fix EPOUT->HPOUT1 Mono Mux routing MAINTAINERS: Update Cirrus Logic Codecs maintainers Rob Herring (1): ASoC: Remove unused 'linux,hdmi-audio' Rohit kumar (8): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass-platform: fix memory leak ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers ASoC: qcom: lpass-cpu: fix concurrency issue ASoC: qcom: lpass-cpu: Use platform_get_resource ASoC: qcom: lpass-platform: Use platform_get_irq ASoC: Add sc7180 lpass cpu node ASoC: lpass-cpu: Move to yaml format Samuel Holland (30): ASoC: dt-bindings: Add a new compatible for the A64 codec ASoC: sun8i-codec: Fix DAPM to match the hardware topology ASoC: sun8i-codec: Add missing mixer routes ASoC: sun8i-codec: Add a quirk for LRCK inversion ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend ASoC: sun50i-codec-analog: Group and sort mixer routes ASoC: sun50i-codec-analog: Make headphone routes stereo ASoC: sun50i-codec-analog: Enable DAPM for headphone switch ASoC: sun50i-codec-analog: Make line out routes stereo ASoC: sun50i-codec-analog: Enable DAPM for line out switch ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch ASoC: sun8i-codec: Hook up component probe function ASoC: sun8i-codec: Remove extraneous widgets ASoC: sun8i-codec: Fix AIF1 MODCLK widget name ASoC: sun8i-codec: Fix AIF1_ADCDAT_CTRL field names ASoC: sun8i-codec: Fix AIF1_MXR_SRC field names ASoC: sun8i-codec: Fix ADC_DIG_CTRL field name ASoC: sun8i-codec: Fix field bit number indentation ASoC: sun8i-codec: Sort masks in a consistent order ASoC: sun8i-codec: Attach the bus clock to the regmap ASoC: sun8i-codec: Manage module clock via DAPM ASoC: sun8i-codec: Set up clock tree at probe time ASoC: sun8i-codec: Swap module clock/reset dependencies ASoC: sun8i-codec: Sort DAPM controls, widgets, and routes ASoC: sun8i-codec: Consistently name DAPM widgets and routes ASoC: sun8i-codec: Correct DAPM widget types ASoC: sun8i-codec: Fix AIF widget channel references ASoC: sun8i-codec: Enable AIF mono/stereo control ASoC: sun8i-codec: Use snd_soc_dai_get_drvdata Sathyanarayana Nujella (3): ASoC: Intel: sof_rt5682: override quirk data for tgl_max98373_rt5682 ASoC: SOF: Add topology filename override based on dmi data match ASoC: SOF: Intel: Use DMI oem string search for tgl_max98373_rt5682 Sebastian Krzyszkowiak (1): ASoC: wm8962: Export DAC/ADC monomix switches Shengjiu Wang (17): ASoC: fsl_sai: Refine enable/disable TE/RE sequence in trigger() ASoC: fsl_sai: Drop TMR/RMR settings for synchronous mode ASoC: fsl_sai: Replace synchronous check with fsl_sai_dir_is_synced ASoC: dt-bindings: ak4458: Add power supply property ASoC: ak4458: Add regulator support ASoC: fsl-asoc-card: Get "extal" clock rate by clk_get_rate ASoC: dt-bindings: ak5558: Add power supply property ASoC: ak5558: Add regulator support ASoC: fsl_sai: Add -EPROBE_DEFER check for regmap init ASoC: fsl_sai: Support multiple data channel enable bits ASoC: fsl_sai: Set SAI Channel Mode to Output Mode ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction ASoC: dt-bindings: ak4458: Add dsd-path property ASoC: ak4458: Add DSD support for ak4458 and ak4497 ASoC: fsl_sai: Instantiate snd_soc_dai_driver Shuming Fan (2): ASoC: rt700: wait for the delayed work to finish when the system suspends ASoC: rt711: wait for the delayed work to finish when the system suspends Srinivas Kandagatla (26): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: q6asm-dai: prepare set params to accept profile change ASoC: q6asm-dai: add gapless support ASoC: q6asm-dai: add support to copy callback ASoC: qcom: add a dedicated menuconfig ASoC: q6dsp: q6afe: add support to Codec DMA ports ASoC: q6dsp: q6routing: add support to Codec DMA ports ASoC: q6dsp: q6afe: prepare afe_apr_send_pkt to take response opcode ASoC: q6dsp: q6afe: add global q6afe waitqueue ASoC: q6dsp: q6afe: add lpass hw voting support ASoC: q6dsp: q6afe: update q6afe_set_param to support global clocks ASoC: q6dsp: q6afe: add codec lpass clocks ASoC: q6dsp: q6afe-dai: add support to Codec DMA ports ASoC: q6afe: dt-bindings: add q6afe clock bindings ASoC: q6afe-clocks: add q6afe clock controller ASoC: qdsp6: add ifdef CONFIG_OF around of_device_id ASoC: q6asm: fix kernel doc warnings ASoC: q6afe-clocks: Fix typo in SPDX Licence regmap: add support to regmap_field_bulk_alloc/free apis ASoC: lpass-platform: use devm_regmap_field_bulk_alloc Stephan Gerhold (5): ASoC: core: Add common helper to parse aux devs from device tree ASoC: simple-card: Use snd_soc_of_parse_aux_devs() ASoC: meson: Use snd_soc_of_parse_aux_devs() ASoC: dt-bindings: qcom: Document "aux-devs" property ASoC: qcom: common: Parse auxiliary devices from device tree Stephen Boyd (3): ASoC: rt5682: Use dev_dbg() in rt5682_clk_check() ASoC: rt5682: Drop usage of __clk_get_name() ASoC: rt5682: Use clk_hw based APIs for registration Takashi Iwai (3): ASoC: fsl: Replace tasklet with work ASoC: sh: Replace tasklet with work ASoC: txx9: Replace tasklet with work Tang Bin (1): ASoC: fsl_spdif: Fix unnecessary check in fsl_spdif_probe() Tuo Li (1): ALSA: rockchip_i2s: fix a possible divide-by-zero bug in rockchip_i2s_hw_params() Tzung-Bi Shih (7): ASoC: mt6359: fix failed to parse DT properties ASoC: rt1015p: add codec driver ASoC: dt-bindings: rt1015p: add document ASoC: dt-bindings: mt8183-da7219: add compatible string for using rt1015p ASoC: mediatek: mt8183-da7219: support machine driver with rt1015p ASoC: mediatek: mt8183-da7219: support jack detection for LINEOUT ASoC: mediatek: mt8183-da7219: fix wrong ops for I2S3 V Sujith Kumar Reddy (8): ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API Asoc: qcom: lpass:Update lpaif_dmactl members order ASoC: qcom: Add support for lpass hdmi driver Asoc: qcom: lpass-platform : Increase buffer size ASoC: qcom: sc7180: Add support for audio over DP Viorel Suman (1): ASoC: fsl_audmix: make clock and output src write only Xu Wang (1): ASoC: fsl: imx-audmix: Use devm_kcalloc() instead of devm_kzalloc() Yong Zhi (1): ASoC: SOF: Intel: hda: Remove unused parameters in cl_dsp_init() Youling Tang (1): sound/soc/intel: Fix spelling mistake "cant" --> "can't" Yu Kuai (1): ASoC: fsl: imx-es8328: add missing put_device() call in imx_es8328_probe() Zhang Qilong (1): ASoC: ti: omap-mcbsp: use devm_platform_ioremap_resource_byname derek.fang (2): ASoC: rt1015: Fix DC calibration on bypass boost mode ASoC: rt1015: Fix the failure to flush DAC data before playback .../ABI/testing/sysfs-bus-pci-devices-catpt | 16 + Documentation/devicetree/bindings/sound/ak4458.txt | 5 + Documentation/devicetree/bindings/sound/ak5558.txt | 2 + .../bindings/sound/allwinner,sun8i-a33-codec.yaml | 6 +- .../devicetree/bindings/sound/cirrus,cs4234.yaml | 74 + .../devicetree/bindings/sound/fsl,spdif.txt | 68 - .../devicetree/bindings/sound/fsl,spdif.yaml | 110 + .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 + Documentation/devicetree/bindings/sound/hdmi.txt | 16 - .../bindings/sound/intel,keembay-i2s.yaml | 1 + .../devicetree/bindings/sound/max98090.txt | 2 +- .../devicetree/bindings/sound/mchp,spdifrx.yaml | 73 + .../devicetree/bindings/sound/mchp,spdiftx.yaml | 75 + .../devicetree/bindings/sound/mt6359.yaml | 61 + .../bindings/sound/mt8183-da7219-max98357.txt | 1 + .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 7 + .../devicetree/bindings/sound/qcom,apq8096.txt | 8 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 - .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 219 ++ .../devicetree/bindings/sound/qcom,q6afe.txt | 23 + .../devicetree/bindings/sound/qcom,sdm845.txt | 8 + .../devicetree/bindings/sound/realtek,rt1015p.yaml | 36 + .../devicetree/bindings/sound/rockchip-spdif.yaml | 1 + Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- .../bindings/sound/samsung,aries-wm8994.yaml | 11 +- .../bindings/sound/samsung,midas-audio.yaml | 6 +- .../devicetree/bindings/sound/samsung,odroid.yaml | 5 + .../devicetree/bindings/sound/samsung-i2s.yaml | 15 + .../devicetree/bindings/sound/sgtl5000.yaml | 4 + .../devicetree/bindings/sound/tas2562.txt | 37 - .../devicetree/bindings/sound/tas2562.yaml | 8 + .../devicetree/bindings/sound/tas2764.yaml | 76 + .../devicetree/bindings/sound/tas2770.yaml | 11 +- .../bindings/sound/ti,j721e-cpb-audio.yaml | 92 +- .../devicetree/bindings/sound/tlv320adcx140.yaml | 50 + MAINTAINERS | 1 + arch/arm/mach-s3c24xx/mach-gta02.c | 17 + arch/arm/mach-s3c24xx/mach-h1940.c | 18 + arch/arm/mach-s3c24xx/mach-rx1950.c | 16 + drivers/base/regmap/regmap.c | 100 + drivers/iio/adc/stm32-dfsdm-adc.c | 4 + drivers/iio/adc/stm32-dfsdm.h | 2 + drivers/spi/spi-atmel.c | 1 - include/dt-bindings/sound/qcom,q6afe.h | 96 +- include/dt-bindings/sound/sc7180-lpass.h | 11 + include/linux/pxa2xx_ssp.h | 4 + include/linux/regmap.h | 11 + include/sound/hda_register.h | 2 +- include/sound/hdaudio_ext.h | 2 +- include/sound/hdmi-codec.h | 3 - include/sound/soc-acpi-intel-match.h | 1 - include/sound/soc-acpi.h | 4 +- include/sound/soc-component.h | 28 +- include/sound/soc-dai.h | 8 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 4 + include/sound/sof.h | 2 + include/sound/sof/ext_manifest.h | 7 + include/sound/sof/info.h | 6 +- include/sound/sof/topology.h | 12 +- include/trace/events/hswadsp.h | 385 --- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 5 + sound/hda/ext/hdac_ext_bus.c | 5 +- sound/soc/amd/acp3x-rt5682-max9836.c | 11 +- sound/soc/atmel/Kconfig | 25 + sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/mchp-spdifrx.c | 953 +++++++ sound/soc/atmel/mchp-spdiftx.c | 871 +++++++ sound/soc/codecs/Kconfig | 27 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ak4458.c | 120 +- sound/soc/codecs/ak4458.h | 5 +- sound/soc/codecs/ak5558.c | 30 + sound/soc/codecs/cs4234.c | 918 +++++++ sound/soc/codecs/cs4234.h | 287 ++ sound/soc/codecs/cs42l51.c | 9 + sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/da7219-aad.c | 85 +- sound/soc/codecs/da7219-aad.h | 3 + sound/soc/codecs/da7219.c | 494 ++-- sound/soc/codecs/da7219.h | 1 + sound/soc/codecs/hdac_hdmi.c | 140 +- sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/codecs/max98373-sdw.c | 3 + sound/soc/codecs/max9867.c | 131 +- sound/soc/codecs/max9867.h | 4 +- sound/soc/codecs/mt6359.c | 2758 ++++++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 +++++++++++++++++++ sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/rt1015.c | 111 +- sound/soc/codecs/rt1015.h | 8 + sound/soc/codecs/rt1015p.c | 148 ++ sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 1 + sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 97 +- sound/soc/codecs/rt5682.h | 2 - sound/soc/codecs/rt700-sdw.c | 5 +- sound/soc/codecs/rt711-sdw.c | 6 +- sound/soc/codecs/rt715-sdw.c | 3 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/rt715.h | 2 + sound/soc/codecs/tas2562.c | 63 +- sound/soc/codecs/tas2764.c | 688 +++++ sound/soc/codecs/tas2764.h | 92 + sound/soc/codecs/tas2770.c | 399 ++- sound/soc/codecs/tas2770.h | 47 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tlv320adcx140.c | 177 +- sound/soc/codecs/tlv320adcx140.h | 8 + sound/soc/codecs/tlv320aic32x4-clk.c | 9 +- sound/soc/codecs/tlv320aic32x4.c | 83 +- sound/soc/codecs/tlv320aic32x4.h | 7 + sound/soc/codecs/wm0010.c | 3 +- sound/soc/codecs/wm8523.h | 2 +- sound/soc/codecs/wm8962.c | 2 + sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 28 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl-asoc-card.c | 15 + sound/soc/fsl/fsl_audmix.c | 16 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_esai.c | 14 +- sound/soc/fsl/fsl_sai.c | 309 ++- sound/soc/fsl/fsl_sai.h | 92 +- sound/soc/fsl/fsl_spdif.c | 4 - sound/soc/fsl/imx-audmix.c | 8 +- sound/soc/fsl/imx-es8328.c | 12 +- sound/soc/fsl/imx-mc13783.c | 10 +- sound/soc/fsl/mx27vis-aic32x4.c | 10 +- sound/soc/generic/simple-card.c | 33 +- sound/soc/intel/Kconfig | 50 +- sound/soc/intel/Makefile | 3 +- sound/soc/intel/atom/sst-atom-controls.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-compress.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/atom/sst-mfld-platform.h | 4 +- sound/soc/intel/atom/sst/sst.c | 6 +- sound/soc/intel/atom/sst/sst.h | 41 +- sound/soc/intel/atom/sst/sst_acpi.c | 1 - sound/soc/intel/atom/sst/sst_drv_interface.c | 3 - sound/soc/intel/atom/sst/sst_ipc.c | 1 - sound/soc/intel/atom/sst/sst_loader.c | 4 +- sound/soc/intel/atom/sst/sst_pvt.c | 5 +- sound/soc/intel/atom/sst/sst_stream.c | 1 - sound/soc/intel/baytrail/Makefile | 5 - sound/soc/intel/baytrail/sst-baytrail-dsp.c | 358 --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 772 ------ sound/soc/intel/baytrail/sst-baytrail-ipc.h | 65 - sound/soc/intel/baytrail/sst-baytrail-pcm.c | 459 ---- sound/soc/intel/boards/Kconfig | 36 +- sound/soc/intel/boards/Makefile | 11 +- sound/soc/intel/boards/bdw-rt5650.c | 47 +- sound/soc/intel/boards/bdw-rt5677.c | 42 +- sound/soc/intel/boards/broadwell.c | 45 +- sound/soc/intel/boards/byt-max98090.c | 182 -- sound/soc/intel/boards/byt-rt5640.c | 224 -- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 1 - sound/soc/intel/boards/bytcr_rt5640.c | 1 - sound/soc/intel/boards/haswell.c | 32 +- sound/soc/intel/boards/hda_dsp_common.c | 7 +- sound/soc/intel/boards/hda_dsp_common.h | 3 - sound/soc/intel/boards/sof_rt5682.c | 13 + sound/soc/intel/boards/sof_sdw.c | 348 ++- sound/soc/intel/boards/sof_sdw_common.h | 26 +- sound/soc/intel/boards/sof_sdw_dmic.c | 1 + sound/soc/intel/boards/sof_sdw_max98373.c | 38 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 6 + sound/soc/intel/boards/sof_sdw_rt1316.c | 119 + sound/soc/intel/boards/sof_sdw_rt5682.c | 2 + sound/soc/intel/boards/sof_sdw_rt700.c | 8 +- sound/soc/intel/boards/sof_sdw_rt711.c | 2 + sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 174 ++ sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 42 + sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 188 ++ sound/soc/intel/catpt/device.c | 355 +++ sound/soc/intel/catpt/dsp.c | 578 ++++ sound/soc/intel/catpt/ipc.c | 298 +++ sound/soc/intel/catpt/loader.c | 671 +++++ sound/soc/intel/catpt/messages.c | 313 +++ sound/soc/intel/catpt/messages.h | 401 +++ sound/soc/intel/catpt/pcm.c | 1175 +++++++++ sound/soc/intel/catpt/registers.h | 178 ++ sound/soc/intel/catpt/sysfs.c | 55 + sound/soc/intel/catpt/trace.h | 83 + sound/soc/intel/common/Makefile | 4 - sound/soc/intel/common/soc-acpi-intel-byt-match.c | 15 - sound/soc/intel/common/soc-acpi-intel-cml-match.c | 89 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 34 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 16 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 232 +- sound/soc/intel/common/sst-acpi.c | 236 -- sound/soc/intel/common/sst-dsp-priv.h | 284 +- sound/soc/intel/common/sst-dsp.c | 162 -- sound/soc/intel/common/sst-dsp.h | 237 +- sound/soc/intel/common/sst-firmware.c | 1273 --------- sound/soc/intel/common/sst-ipc.c | 27 - sound/soc/intel/common/sst-ipc.h | 3 - sound/soc/intel/haswell/Makefile | 5 - sound/soc/intel/haswell/sst-haswell-dsp.c | 705 ----- sound/soc/intel/haswell/sst-haswell-ipc.c | 2222 ---------------- sound/soc/intel/haswell/sst-haswell-ipc.h | 527 ---- sound/soc/intel/haswell/sst-haswell-pcm.c | 1369 ---------- sound/soc/intel/keembay/kmb_platform.c | 145 +- sound/soc/intel/skylake/bxt-sst.c | 2 - sound/soc/intel/skylake/cnl-sst-dsp.h | 4 +- sound/soc/intel/skylake/cnl-sst.c | 4 +- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/intel/skylake/skl-sst-cldma.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 16 +- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-sst.c | 4 +- sound/soc/intel/skylake/skl-topology.c | 8 +- sound/soc/intel/skylake/skl-topology.h | 8 +- sound/soc/intel/skylake/skl.c | 10 +- sound/soc/intel/skylake/skl.h | 2 +- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 50 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/meson/axg-tdm-formatter.c | 2 +- sound/soc/meson/meson-card-utils.c | 33 +- sound/soc/qcom/Kconfig | 24 +- sound/soc/qcom/Makefile | 4 + sound/soc/qcom/common.c | 4 + sound/soc/qcom/lpass-apq8016.c | 92 +- sound/soc/qcom/lpass-cpu.c | 463 +++- sound/soc/qcom/lpass-hdmi.c | 258 ++ sound/soc/qcom/lpass-hdmi.h | 102 + sound/soc/qcom/lpass-ipq806x.c | 71 +- sound/soc/qcom/lpass-lpaif-reg.h | 208 +- sound/soc/qcom/lpass-platform.c | 493 +++- sound/soc/qcom/lpass-sc7180.c | 305 +++ sound/soc/qcom/lpass.h | 181 +- sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6adm.c | 2 + sound/soc/qcom/qdsp6/q6afe-clocks.c | 272 ++ sound/soc/qcom/qdsp6/q6afe-dai.c | 231 ++ sound/soc/qcom/qdsp6/q6afe.c | 310 ++- sound/soc/qcom/qdsp6/q6afe.h | 33 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 416 ++- sound/soc/qcom/qdsp6/q6asm.c | 175 +- sound/soc/qcom/qdsp6/q6asm.h | 49 +- sound/soc/qcom/qdsp6/q6core.c | 2 + sound/soc/qcom/qdsp6/q6routing.c | 123 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 3 +- sound/soc/samsung/h1940_uda1380.c | 71 +- sound/soc/samsung/neo1973_wm8753.c | 85 +- sound/soc/samsung/rx1950_uda1380.c | 72 +- sound/soc/samsung/s3c-i2s-v2.c | 3 +- sound/soc/samsung/s3c-i2s-v2.h | 3 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_pcm.c | 21 +- sound/soc/soc-component.c | 73 +- sound/soc/soc-compress.c | 30 +- sound/soc/soc-core.c | 76 +- sound/soc/soc-dai.c | 21 +- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 5 + sound/soc/soc-link.c | 21 +- sound/soc/soc-pcm.c | 123 +- sound/soc/soc-topology.c | 18 +- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/control.c | 149 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/imx/Kconfig | 10 + sound/soc/sof/imx/Makefile | 3 + sound/soc/sof/imx/imx-common.c | 75 + sound/soc/sof/imx/imx-common.h | 16 + sound/soc/sof/imx/imx8.c | 40 +- sound/soc/sof/imx/imx8m.c | 27 +- sound/soc/sof/intel/Kconfig | 31 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 6 +- sound/soc/sof/intel/cnl.c | 40 +- sound/soc/sof/intel/hda-codec.c | 28 +- sound/soc/sof/intel/hda-dsp.c | 20 +- sound/soc/sof/intel/hda-ipc.h | 4 + sound/soc/sof/intel/hda-loader.c | 196 +- sound/soc/sof/intel/hda-stream.c | 69 + sound/soc/sof/intel/hda.c | 47 +- sound/soc/sof/intel/hda.h | 12 +- sound/soc/sof/intel/shim.h | 2 +- sound/soc/sof/intel/tgl.c | 153 ++ sound/soc/sof/ipc.c | 16 +- sound/soc/sof/loader.c | 34 +- sound/soc/sof/pcm.c | 8 +- sound/soc/sof/pm.c | 13 +- sound/soc/sof/sof-acpi-dev.c | 2 + sound/soc/sof/sof-audio.c | 56 +- sound/soc/sof/sof-audio.h | 10 + sound/soc/sof/sof-of-dev.c | 6 + sound/soc/sof/sof-pci-dev.c | 59 +- sound/soc/sof/sof-priv.h | 22 +- sound/soc/sof/topology.c | 421 +-- sound/soc/sprd/sprd-pcm-compress.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 8 +- sound/soc/stm/stm32_sai_sub.c | 10 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun50i-codec-analog.c | 176 +- sound/soc/sunxi/sun8i-codec.c | 473 ++-- sound/soc/tegra/tegra_max98090.c | 12 +- sound/soc/tegra/tegra_rt5640.c | 12 +- sound/soc/tegra/tegra_wm8753.c | 12 +- sound/soc/tegra/trimslice.c | 12 +- sound/soc/ti/Kconfig | 2 +- sound/soc/ti/davinci-mcasp.c | 4 +- sound/soc/ti/j721e-evm.c | 13 +- sound/soc/txx9/txx9aclc.c | 11 +- sound/soc/txx9/txx9aclc.h | 2 +- sound/soc/xilinx/Kconfig | 4 +- 327 files changed, 23460 insertions(+), 12639 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-catpt create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4234.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2764.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h delete mode 100644 include/trace/events/hswadsp.h create mode 100644 sound/soc/atmel/mchp-spdifrx.c create mode 100644 sound/soc/atmel/mchp-spdiftx.c create mode 100644 sound/soc/codecs/cs4234.c create mode 100644 sound/soc/codecs/cs4234.h create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h create mode 100644 sound/soc/codecs/rt1015p.c create mode 100644 sound/soc/codecs/tas2764.c create mode 100644 sound/soc/codecs/tas2764.h delete mode 100644 sound/soc/intel/baytrail/Makefile delete mode 100644 sound/soc/intel/baytrail/sst-baytrail-dsp.c delete mode 100644 sound/soc/intel/baytrail/sst-baytrail-ipc.c delete mode 100644 sound/soc/intel/baytrail/sst-baytrail-ipc.h delete mode 100644 sound/soc/intel/baytrail/sst-baytrail-pcm.c delete mode 100644 sound/soc/intel/boards/byt-max98090.c delete mode 100644 sound/soc/intel/boards/byt-rt5640.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1316.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711_sdca.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/sysfs.c create mode 100644 sound/soc/intel/catpt/trace.h delete mode 100644 sound/soc/intel/common/sst-acpi.c delete mode 100644 sound/soc/intel/common/sst-firmware.c delete mode 100644 sound/soc/intel/haswell/Makefile delete mode 100644 sound/soc/intel/haswell/sst-haswell-dsp.c delete mode 100644 sound/soc/intel/haswell/sst-haswell-ipc.c delete mode 100644 sound/soc/intel/haswell/sst-haswell-ipc.h delete mode 100644 sound/soc/intel/haswell/sst-haswell-pcm.c create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h create mode 100644 sound/soc/qcom/lpass-sc7180.c create mode 100644 sound/soc/qcom/qdsp6/q6afe-clocks.c create mode 100644 sound/soc/sof/imx/imx-common.c create mode 100644 sound/soc/sof/imx/imx-common.h create mode 100644 sound/soc/sof/intel/tgl.c

4 9

bug in the ALSA driver 'snd_hda_intel'
by GitHub issues - opened 21 Oct '20

21 Oct '20

alsa-project/alsa-lib issue #90 was opened from Mysgym: Hello, after spending some time troubleshooting an issue with an audio interface (using snd_hda_intel). I've stumbled upon this message from PulseAudio in the journal : `Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 17 18:30:58 LaptopName pulseaudio[1782]: E: [alsa-sink-ALC256 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ` So I filed a bug report This is Manjaro on Kernel 5.8 Issue URL : https://github.com/alsa-project/alsa-lib/issues/90 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: Intel: sof_rt5682: update quirk for cml boards
by Brent Lu 21 Oct '20

21 Oct '20

The default quirk data of sof_rt5682 is for tgl platform. For cml platforms to reuse this driver, the flag SOF_RT5682_MCLK_24MHZ is necessary to setup codec asrc correctly. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ddbb9fe7cc06..fa2c226a444a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -768,6 +768,11 @@ static int sof_audio_probe(struct platform_device *pdev) /* default number of HDMI DAI's */ if (!hdmi_num) hdmi_num = 3; + + if (soc_intel_is_cml()) { + /* default quirk for sof_rt5682 is for tgl platform */ + sof_rt5682_quirk |= SOF_RT5682_MCLK_24MHZ; + } } /* need to get main clock from pmc */ -- 2.17.1

3 2

[PATCH v2] ASoC: amd: move the call to devm_kzalloc below platform_get_resource()
by Defang Bo 21 Oct '20

21 Oct '20

Just as the commit <4cb79ef9c6c4>("ASoC: amd: Fix potential NULL pointer dereference"),it makes no sense to allocate any resources if platform_get_resource fails,so move the call to devm_kzalloc() below the mentioned code. Signed-off-by: Defang Bo <bodefang(a)126.com> --- sound/soc/amd/raven/acp3x-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index c9c69eb..04ec603 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -307,9 +307,11 @@ static int acp3x_dai_probe(struct platform_device *pdev) dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n"); return -ENOMEM; } + adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), GFP_KERNEL); if (!adata) return -ENOMEM; + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (!adata->acp3x_base) -- 1.9.1

2 1

[RFC] treewide: cleanup unreachable breaks
by trix＠redhat.com 20 Oct '20

20 Oct '20

From: Tom Rix <trix(a)redhat.com> This is a upcoming change to clean up a new warning treewide. I am wondering if the change could be one mega patch (see below) or normal patch per file about 100 patches or somewhere half way by collecting early acks. clang has a number of useful, new warnings see https://clang.llvm.org/docs/DiagnosticsReference.html This change cleans up -Wunreachable-code-break https://clang.llvm.org/docs/DiagnosticsReference.html#wunreachable-code-bre… for 266 of 485 warnings in this week's linux-next, allyesconfig on x86_64. The method of fixing was to look for warnings where the preceding statement was a simple statement and by inspection made the subsequent break unneeded. In order of frequency these look like return and break switch (c->x86_vendor) { case X86_VENDOR_INTEL: intel_p5_mcheck_init(c); return 1; - break; goto and break default: operation = 0; /* make gcc happy */ goto fail_response; - break; break and break case COLOR_SPACE_SRGB: /* by pass */ REG_SET(OUTPUT_CSC_CONTROL, 0, OUTPUT_CSC_GRPH_MODE, 0); break; - break; The exception to the simple statement, is a switch case with a block and the end of block is a return struct obj_buffer *buff = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "size=%u\naddr=0x%X\n", buff->size, buff->addr); } - break; Not considered obvious and excluded, breaks after multi level switches complicated if-else if-else blocks panic() or similar calls And there is an odd addition of a 'fallthrough' in drivers/tty/nozomi.c Tom --- diff --git a/arch/x86/kernel/cpu/mce/core.c b/arch/x86/kernel/cpu/mce/core.c index 1c08cb9eb9f6..16ce86aed8e2 100644 --- a/arch/x86/kernel/cpu/mce/core.c +++ b/arch/x86/kernel/cpu/mce/core.c @@ -1809,15 +1809,13 @@ static int __mcheck_cpu_ancient_init(struct cpuinfo_x86 *c) switch (c->x86_vendor) { case X86_VENDOR_INTEL: intel_p5_mcheck_init(c); return 1; - break; case X86_VENDOR_CENTAUR: winchip_mcheck_init(c); return 1; - break; default: return 0; } return 0; diff --git a/arch/x86/kernel/cpu/microcode/amd.c b/arch/x86/kernel/cpu/microcode/amd.c index 3f6b137ef4e6..3d4a48336084 100644 --- a/arch/x86/kernel/cpu/microcode/amd.c +++ b/arch/x86/kernel/cpu/microcode/amd.c @@ -213,11 +213,10 @@ static unsigned int __verify_patch_size(u8 family, u32 sh_psize, size_t buf_size max_size = F14H_MPB_MAX_SIZE; break; default: WARN(1, "%s: WTF family: 0x%x\n", __func__, family); return 0; - break; } if (sh_psize > min_t(u32, buf_size, max_size)) return 0; diff --git a/drivers/acpi/utils.c b/drivers/acpi/utils.c index 838b719ec7ce..d5411a166685 100644 --- a/drivers/acpi/utils.c +++ b/drivers/acpi/utils.c @@ -102,11 +102,10 @@ acpi_extract_package(union acpi_object *package, printk(KERN_WARNING PREFIX "Invalid package element" " [%d]: got number, expecting" " [%c]\n", i, format_string[i]); return AE_BAD_DATA; - break; } break; case ACPI_TYPE_STRING: case ACPI_TYPE_BUFFER: @@ -127,11 +126,10 @@ acpi_extract_package(union acpi_object *package, printk(KERN_WARNING PREFIX "Invalid package element" " [%d] got string/buffer," " expecting [%c]\n", i, format_string[i]); return AE_BAD_DATA; - break; } break; case ACPI_TYPE_LOCAL_REFERENCE: switch (format_string[i]) { case 'R': @@ -142,22 +140,20 @@ acpi_extract_package(union acpi_object *package, printk(KERN_WARNING PREFIX "Invalid package element" " [%d] got reference," " expecting [%c]\n", i, format_string[i]); return AE_BAD_DATA; - break; } break; case ACPI_TYPE_PACKAGE: default: ACPI_DEBUG_PRINT((ACPI_DB_INFO, "Found unsupported element at index=%d\n", i)); /* TBD: handle nested packages... */ return AE_SUPPORT; - break; } } /* * Validate output buffer. diff --git a/drivers/base/power/main.c b/drivers/base/power/main.c index 205a06752ca9..c7ac49042cee 100644 --- a/drivers/base/power/main.c +++ b/drivers/base/power/main.c @@ -361,11 +361,10 @@ static pm_callback_t pm_op(const struct dev_pm_ops *ops, pm_message_t state) case PM_EVENT_HIBERNATE: return ops->poweroff; case PM_EVENT_THAW: case PM_EVENT_RECOVER: return ops->thaw; - break; case PM_EVENT_RESTORE: return ops->restore; #endif /* CONFIG_HIBERNATE_CALLBACKS */ } diff --git a/drivers/block/xen-blkback/blkback.c b/drivers/block/xen-blkback/blkback.c index adfc9352351d..f769fbd1b4c4 100644 --- a/drivers/block/xen-blkback/blkback.c +++ b/drivers/block/xen-blkback/blkback.c @@ -1267,11 +1267,10 @@ static int dispatch_rw_block_io(struct xen_blkif_ring *ring, operation_flags = REQ_PREFLUSH; break; default: operation = 0; /* make gcc happy */ goto fail_response; - break; } /* Check that the number of segments is sane. */ nseg = req->operation == BLKIF_OP_INDIRECT ? req->u.indirect.nr_segments : req->u.rw.nr_segments; diff --git a/drivers/char/ipmi/ipmi_devintf.c b/drivers/char/ipmi/ipmi_devintf.c index f7b1c004a12b..3dd1d5abb298 100644 --- a/drivers/char/ipmi/ipmi_devintf.c +++ b/drivers/char/ipmi/ipmi_devintf.c @@ -488,11 +488,10 @@ static long ipmi_ioctl(struct file *file, rv = -EFAULT; break; } return ipmi_set_my_address(priv->user, val.channel, val.value); - break; } case IPMICTL_GET_MY_CHANNEL_ADDRESS_CMD: { struct ipmi_channel_lun_address_set val; diff --git a/drivers/char/lp.c b/drivers/char/lp.c index 0ec73917d8dd..862c2fd933c7 100644 --- a/drivers/char/lp.c +++ b/drivers/char/lp.c @@ -620,11 +620,10 @@ static int lp_do_ioctl(unsigned int minor, unsigned int cmd, case LPWAIT: LP_WAIT(minor) = arg; break; case LPSETIRQ: return -EINVAL; - break; case LPGETIRQ: if (copy_to_user(argp, &LP_IRQ(minor), sizeof(int))) return -EFAULT; break; diff --git a/drivers/char/mwave/mwavedd.c b/drivers/char/mwave/mwavedd.c index e43c876a9223..11272d605ecd 100644 --- a/drivers/char/mwave/mwavedd.c +++ b/drivers/char/mwave/mwavedd.c @@ -401,11 +401,10 @@ static long mwave_ioctl(struct file *file, unsigned int iocmd, } break; default: return -ENOTTY; - break; } /* switch */ PRINTK_2(TRACE_MWAVE, "mwavedd::mwave_ioctl, exit retval %x\n", retval); return retval; diff --git a/drivers/crypto/atmel-sha.c b/drivers/crypto/atmel-sha.c index 75ccf41a7cb9..0eb6f54e3b66 100644 --- a/drivers/crypto/atmel-sha.c +++ b/drivers/crypto/atmel-sha.c @@ -457,11 +457,10 @@ static int atmel_sha_init(struct ahash_request *req) ctx->flags |= SHA_FLAGS_SHA512; ctx->block_size = SHA512_BLOCK_SIZE; break; default: return -EINVAL; - break; } ctx->bufcnt = 0; ctx->digcnt[0] = 0; ctx->digcnt[1] = 0; diff --git a/drivers/edac/amd64_edac.c b/drivers/edac/amd64_edac.c index fcc08bbf6945..386a3a4cf279 100644 --- a/drivers/edac/amd64_edac.c +++ b/drivers/edac/amd64_edac.c @@ -2459,38 +2459,30 @@ static int map_err_sym_to_channel(int err_sym, int sym_size) if (sym_size == 4) switch (err_sym) { case 0x20: case 0x21: return 0; - break; case 0x22: case 0x23: return 1; - break; default: return err_sym >> 4; - break; } /* x8 symbols */ else switch (err_sym) { /* imaginary bits not in a DIMM */ case 0x10: WARN(1, KERN_ERR "Invalid error symbol: 0x%x\n", err_sym); return -1; - break; - case 0x11: return 0; - break; case 0x12: return 1; - break; default: return err_sym >> 3; - break; } return -1; } static int get_channel_from_ecc_syndrome(struct mem_ctl_info *mci, u16 syndrome) diff --git a/drivers/gpio/gpio-bd70528.c b/drivers/gpio/gpio-bd70528.c index 45b3da8da336..931e5765fe92 100644 --- a/drivers/gpio/gpio-bd70528.c +++ b/drivers/gpio/gpio-bd70528.c @@ -69,21 +69,18 @@ static int bd70528_gpio_set_config(struct gpio_chip *chip, unsigned int offset, case PIN_CONFIG_DRIVE_OPEN_DRAIN: return regmap_update_bits(bdgpio->chip.regmap, GPIO_OUT_REG(offset), BD70528_GPIO_DRIVE_MASK, BD70528_GPIO_OPEN_DRAIN); - break; case PIN_CONFIG_DRIVE_PUSH_PULL: return regmap_update_bits(bdgpio->chip.regmap, GPIO_OUT_REG(offset), BD70528_GPIO_DRIVE_MASK, BD70528_GPIO_PUSH_PULL); - break; case PIN_CONFIG_INPUT_DEBOUNCE: return bd70528_set_debounce(bdgpio, offset, pinconf_to_config_argument(config)); - break; default: break; } return -ENOTSUPP; } diff --git a/drivers/gpu/drm/amd/display/dc/dce/dce_transform.c b/drivers/gpu/drm/amd/display/dc/dce/dce_transform.c index 2a32b66959ba..130a0a0c8332 100644 --- a/drivers/gpu/drm/amd/display/dc/dce/dce_transform.c +++ b/drivers/gpu/drm/amd/display/dc/dce/dce_transform.c @@ -1328,11 +1328,10 @@ static bool configure_graphics_mode( case COLOR_SPACE_SRGB: /* by pass */ REG_SET(OUTPUT_CSC_CONTROL, 0, OUTPUT_CSC_GRPH_MODE, 0); break; - break; case COLOR_SPACE_SRGB_LIMITED: /* TV RGB */ REG_SET(OUTPUT_CSC_CONTROL, 0, OUTPUT_CSC_GRPH_MODE, 1); break; diff --git a/drivers/gpu/drm/amd/display/dc/dce100/dce100_resource.c b/drivers/gpu/drm/amd/display/dc/dce100/dce100_resource.c index d741787f75dc..42c7d157da32 100644 --- a/drivers/gpu/drm/amd/display/dc/dce100/dce100_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce100/dce100_resource.c @@ -416,29 +416,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/amd/display/dc/dce110/dce110_resource.c b/drivers/gpu/drm/amd/display/dc/dce110/dce110_resource.c index 2bbfa2e176a9..382581c4a674 100644 --- a/drivers/gpu/drm/amd/display/dc/dce110/dce110_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce110/dce110_resource.c @@ -469,29 +469,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/amd/display/dc/dce112/dce112_resource.c b/drivers/gpu/drm/amd/display/dc/dce112/dce112_resource.c index b622b4b1dac3..7b4b2304bbff 100644 --- a/drivers/gpu/drm/amd/display/dc/dce112/dce112_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce112/dce112_resource.c @@ -444,29 +444,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/amd/display/dc/dce120/dce120_resource.c b/drivers/gpu/drm/amd/display/dc/dce120/dce120_resource.c index 16fe7344702f..3d782b7c86cb 100644 --- a/drivers/gpu/drm/amd/display/dc/dce120/dce120_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce120/dce120_resource.c @@ -381,29 +381,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/amd/display/dc/dce60/dce60_resource.c b/drivers/gpu/drm/amd/display/dc/dce60/dce60_resource.c index 5a5a9cb77acb..e9dd78c484d6 100644 --- a/drivers/gpu/drm/amd/display/dc/dce60/dce60_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce60/dce60_resource.c @@ -451,29 +451,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/amd/display/dc/dce80/dce80_resource.c b/drivers/gpu/drm/amd/display/dc/dce80/dce80_resource.c index 0eae8cd35f9a..9dbf658162cd 100644 --- a/drivers/gpu/drm/amd/display/dc/dce80/dce80_resource.c +++ b/drivers/gpu/drm/amd/display/dc/dce80/dce80_resource.c @@ -456,29 +456,22 @@ static int map_transmitter_id_to_phy_instance( enum transmitter transmitter) { switch (transmitter) { case TRANSMITTER_UNIPHY_A: return 0; - break; case TRANSMITTER_UNIPHY_B: return 1; - break; case TRANSMITTER_UNIPHY_C: return 2; - break; case TRANSMITTER_UNIPHY_D: return 3; - break; case TRANSMITTER_UNIPHY_E: return 4; - break; case TRANSMITTER_UNIPHY_F: return 5; - break; case TRANSMITTER_UNIPHY_G: return 6; - break; default: ASSERT(0); return 0; } } diff --git a/drivers/gpu/drm/mgag200/mgag200_mode.c b/drivers/gpu/drm/mgag200/mgag200_mode.c index 38672f9e5c4f..bbe4e60dfd08 100644 --- a/drivers/gpu/drm/mgag200/mgag200_mode.c +++ b/drivers/gpu/drm/mgag200/mgag200_mode.c @@ -792,25 +792,20 @@ static int mgag200_crtc_set_plls(struct mga_device *mdev, long clock) case G200_AGP: return mgag200_g200_set_plls(mdev, clock); case G200_SE_A: case G200_SE_B: return mga_g200se_set_plls(mdev, clock); - break; case G200_WB: case G200_EW3: return mga_g200wb_set_plls(mdev, clock); - break; case G200_EV: return mga_g200ev_set_plls(mdev, clock); - break; case G200_EH: case G200_EH3: return mga_g200eh_set_plls(mdev, clock); - break; case G200_ER: return mga_g200er_set_plls(mdev, clock); - break; } misc = RREG8(MGA_MISC_IN); misc &= ~MGAREG_MISC_CLK_SEL_MASK; misc |= MGAREG_MISC_CLK_SEL_MGA_MSK; diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/bios/pll.c b/drivers/gpu/drm/nouveau/nvkm/subdev/bios/pll.c index 350f10a3de37..2ec84b8a3b3a 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/bios/pll.c +++ b/drivers/gpu/drm/nouveau/nvkm/subdev/bios/pll.c @@ -121,11 +121,10 @@ pll_map(struct nvkm_bios *bios) case NV_10: case NV_11: case NV_20: case NV_30: return nv04_pll_mapping; - break; case NV_40: return nv40_pll_mapping; case NV_50: if (device->chipset == 0x50) return nv50_pll_mapping; diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/clk/mcp77.c b/drivers/gpu/drm/nouveau/nvkm/subdev/clk/mcp77.c index efa50274df97..4884eb4a9221 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/clk/mcp77.c +++ b/drivers/gpu/drm/nouveau/nvkm/subdev/clk/mcp77.c @@ -138,21 +138,18 @@ mcp77_clk_read(struct nvkm_clk *base, enum nv_clk_src src) case 0x00000030: return read_pll(clk, 0x004020) >> P; } break; case nv_clk_src_mem: return 0; - break; case nv_clk_src_vdec: P = (read_div(clk) & 0x00000700) >> 8; switch (mast & 0x00400000) { case 0x00400000: return nvkm_clk_read(&clk->base, nv_clk_src_core) >> P; - break; default: return 500000 >> P; - break; } break; default: break; } diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/fb/ramnv50.c b/drivers/gpu/drm/nouveau/nvkm/subdev/fb/ramnv50.c index 2ccb4b6be153..7b1eb44ff3da 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/fb/ramnv50.c +++ b/drivers/gpu/drm/nouveau/nvkm/subdev/fb/ramnv50.c @@ -169,11 +169,10 @@ nv50_ram_timing_read(struct nv50_ram *ram, u32 *timing) case NVKM_RAM_TYPE_GDDR3: T(CWL) = ((timing[2] & 0xff000000) >> 24) + 1; break; default: return -ENOSYS; - break; } T(WR) = ((timing[1] >> 24) & 0xff) - 1 - T(CWL); return 0; diff --git a/drivers/gpu/drm/nouveau/nvkm/subdev/top/gk104.c b/drivers/gpu/drm/nouveau/nvkm/subdev/top/gk104.c index e01746ce9fc4..1156634533f9 100644 --- a/drivers/gpu/drm/nouveau/nvkm/subdev/top/gk104.c +++ b/drivers/gpu/drm/nouveau/nvkm/subdev/top/gk104.c @@ -88,11 +88,10 @@ gk104_top_oneinit(struct nvkm_top *top) case 0x0000000e: B_(NVENC ); break; case 0x0000000f: A_(NVENC1); break; case 0x00000010: B_(NVDEC ); break; case 0x00000013: B_(CE ); break; case 0x00000014: C_(GSP ); break; - break; default: break; } nvkm_debug(subdev, "%02x.%d (%8s): addr %06x fault %2d " diff --git a/drivers/gpu/drm/qxl/qxl_ioctl.c b/drivers/gpu/drm/qxl/qxl_ioctl.c index 5cea6eea72ab..2072ddc9549c 100644 --- a/drivers/gpu/drm/qxl/qxl_ioctl.c +++ b/drivers/gpu/drm/qxl/qxl_ioctl.c @@ -158,11 +158,10 @@ static int qxl_process_single_command(struct qxl_device *qdev, case QXL_CMD_SURFACE: case QXL_CMD_CURSOR: default: DRM_DEBUG("Only draw commands in execbuffers\n"); return -EINVAL; - break; } if (cmd->command_size > PAGE_SIZE - sizeof(union qxl_release_info)) return -EINVAL; diff --git a/drivers/iio/adc/meson_saradc.c b/drivers/iio/adc/meson_saradc.c index e03988698755..66dc452d643a 100644 --- a/drivers/iio/adc/meson_saradc.c +++ b/drivers/iio/adc/meson_saradc.c @@ -591,17 +591,15 @@ static int meson_sar_adc_iio_info_read_raw(struct iio_dev *indio_dev, switch (mask) { case IIO_CHAN_INFO_RAW: return meson_sar_adc_get_sample(indio_dev, chan, NO_AVERAGING, ONE_SAMPLE, val); - break; case IIO_CHAN_INFO_AVERAGE_RAW: return meson_sar_adc_get_sample(indio_dev, chan, MEAN_AVERAGING, EIGHT_SAMPLES, val); - break; case IIO_CHAN_INFO_SCALE: if (chan->type == IIO_VOLTAGE) { ret = regulator_get_voltage(priv->vref); if (ret < 0) { diff --git a/drivers/iio/imu/bmi160/bmi160_core.c b/drivers/iio/imu/bmi160/bmi160_core.c index 222ebb26f013..431076dc0d2c 100644 --- a/drivers/iio/imu/bmi160/bmi160_core.c +++ b/drivers/iio/imu/bmi160/bmi160_core.c @@ -484,11 +484,10 @@ static int bmi160_write_raw(struct iio_dev *indio_dev, switch (mask) { case IIO_CHAN_INFO_SCALE: return bmi160_set_scale(data, bmi160_to_sensor(chan->type), val2); - break; case IIO_CHAN_INFO_SAMP_FREQ: return bmi160_set_odr(data, bmi160_to_sensor(chan->type), val, val2); default: return -EINVAL; diff --git a/drivers/ipack/devices/ipoctal.c b/drivers/ipack/devices/ipoctal.c index d480a514c983..3940714e4397 100644 --- a/drivers/ipack/devices/ipoctal.c +++ b/drivers/ipack/devices/ipoctal.c @@ -542,11 +542,10 @@ static void ipoctal_set_termios(struct tty_struct *tty, mr1 |= MR1_RxRTS_CONTROL_OFF; mr2 |= MR2_TxRTS_CONTROL_ON | MR2_CTS_ENABLE_TX_OFF; break; default: return; - break; } baud = tty_get_baud_rate(tty); tty_termios_encode_baud_rate(&tty->termios, baud, baud); diff --git a/drivers/media/dvb-frontends/drx39xyj/drxj.c b/drivers/media/dvb-frontends/drx39xyj/drxj.c index 237b9d04c076..37b32d9b398d 100644 --- a/drivers/media/dvb-frontends/drx39xyj/drxj.c +++ b/drivers/media/dvb-frontends/drx39xyj/drxj.c @@ -2323,11 +2323,10 @@ hi_command(struct i2c_device_addr *dev_addr, const struct drxj_hi_cmd *cmd, u16 /* No parameters */ break; default: return -EINVAL; - break; } /* Write command */ rc = drxj_dap_write_reg16(dev_addr, SIO_HI_RA_RAM_CMD__A, cmd->cmd, 0); if (rc != 0) { @@ -3592,11 +3591,10 @@ static int ctrl_set_uio_cfg(struct drx_demod_instance *demod, struct drxuio_cfg goto rw_error; } break; default: return -EINVAL; - break; } /* switch ( uio_cfg->mode ) */ break; /*====================================================================*/ case DRX_UIO3: /* DRX_UIO3: GPIO UIO-3 */ @@ -3616,11 +3614,10 @@ static int ctrl_set_uio_cfg(struct drx_demod_instance *demod, struct drxuio_cfg goto rw_error; } break; default: return -EINVAL; - break; } /* switch ( uio_cfg->mode ) */ break; /*====================================================================*/ case DRX_UIO4: /* DRX_UIO4: IRQN UIO-4 */ @@ -3640,11 +3637,10 @@ static int ctrl_set_uio_cfg(struct drx_demod_instance *demod, struct drxuio_cfg ext_attr->uio_irqn_mode = uio_cfg->mode; break; case DRX_UIO_MODE_FIRMWARE0: default: return -EINVAL; - break; } /* switch ( uio_cfg->mode ) */ break; /*====================================================================*/ default: return -EINVAL; @@ -10951,11 +10947,10 @@ ctrl_set_standard(struct drx_demod_instance *demod, enum drx_standard *standard) } break; default: ext_attr->standard = DRX_STANDARD_UNKNOWN; return -EINVAL; - break; } return 0; rw_error: /* Don't know what the standard is now ... try again */ @@ -11072,11 +11067,10 @@ ctrl_power_mode(struct drx_demod_instance *demod, enum drx_power_mode *mode) sio_cc_pwd_mode = SIO_CC_PWD_MODE_LEVEL_OSC; break; default: /* Unknow sleep mode */ return -EINVAL; - break; } /* Check if device needs to be powered up */ if ((common_attr->current_power_mode != DRX_POWER_UP)) { rc = power_up_device(demod); @@ -11894,11 +11888,10 @@ static int drx_ctrl_u_code(struct drx_demod_instance *demod, } break; } default: return -EINVAL; - break; } mc_data += mc_block_nr_bytes; } diff --git a/drivers/media/dvb-frontends/drxd_hard.c b/drivers/media/dvb-frontends/drxd_hard.c index 45f982863904..a7eb81df88c2 100644 --- a/drivers/media/dvb-frontends/drxd_hard.c +++ b/drivers/media/dvb-frontends/drxd_hard.c @@ -1620,11 +1620,10 @@ static int CorrectSysClockDeviation(struct drxd_state *state) case 6000000: bandwidth = DRXD_BANDWIDTH_6MHZ_IN_HZ; break; default: return -1; - break; } /* Compute new sysclock value sysClockFreq = (((incr + 2^23)*bandwidth)/2^21)/1000 */ incr += (1 << 23); diff --git a/drivers/media/dvb-frontends/nxt200x.c b/drivers/media/dvb-frontends/nxt200x.c index 35b83b1dd82c..200b6dbc75f8 100644 --- a/drivers/media/dvb-frontends/nxt200x.c +++ b/drivers/media/dvb-frontends/nxt200x.c @@ -166,11 +166,10 @@ static int nxt200x_writereg_multibyte (struct nxt200x_state* state, u8 reg, u8* len2 = ((attr << 4) | 0x10) | len; buf = 0x80; break; default: return -EINVAL; - break; } /* set multi register length */ nxt200x_writebytes(state, 0x34, &len2, 1); @@ -188,11 +187,10 @@ static int nxt200x_writereg_multibyte (struct nxt200x_state* state, u8 reg, u8* if (buf == 0) return 0; break; default: return -EINVAL; - break; } pr_warn("Error writing multireg register 0x%02X\n", reg); return 0; @@ -214,11 +212,10 @@ static int nxt200x_readreg_multibyte (struct nxt200x_state* state, u8 reg, u8* d nxt200x_writebytes(state, 0x34, &len2, 1); /* read the actual data */ nxt200x_readbytes(state, reg, data, len); return 0; - break; case NXT2004: /* probably not right, but gives correct values */ attr = 0x02; if (reg & 0x80) { attr = attr << 1; @@ -237,14 +234,12 @@ static int nxt200x_readreg_multibyte (struct nxt200x_state* state, u8 reg, u8* d /* read the actual data */ for(i = 0; i < len; i++) { nxt200x_readbytes(state, 0x36 + i, &data[i], 1); } return 0; - break; default: return -EINVAL; - break; } } static void nxt200x_microcontroller_stop (struct nxt200x_state* state) { @@ -372,11 +367,10 @@ static int nxt200x_writetuner (struct nxt200x_state* state, u8* data) } pr_warn("timeout error writing to tuner\n"); break; default: return -EINVAL; - break; } return 0; } static void nxt200x_agc_reset(struct nxt200x_state* state) @@ -553,11 +547,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) if (state->config->set_ts_params) state->config->set_ts_params(fe, 0); break; default: return -EINVAL; - break; } if (fe->ops.tuner_ops.calc_regs) { /* get tuning information */ fe->ops.tuner_ops.calc_regs(fe, buf, 5); @@ -578,11 +571,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case VSB_8: buf[0] = 0x70; break; default: return -EINVAL; - break; } nxt200x_writebytes(state, 0x42, buf, 1); /* configure sdm */ switch (state->demod_chip) { @@ -592,11 +584,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case NXT2004: buf[0] = 0x07; break; default: return -EINVAL; - break; } nxt200x_writebytes(state, 0x57, buf, 1); /* write sdm1 input */ buf[0] = 0x10; @@ -608,11 +599,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case NXT2004: nxt200x_writebytes(state, 0x58, buf, 2); break; default: return -EINVAL; - break; } /* write sdmx input */ switch (p->modulation) { case QAM_64: @@ -624,11 +614,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case VSB_8: buf[0] = 0x60; break; default: return -EINVAL; - break; } buf[1] = 0x00; switch (state->demod_chip) { case NXT2002: nxt200x_writereg_multibyte(state, 0x5C, buf, 2); @@ -636,11 +625,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case NXT2004: nxt200x_writebytes(state, 0x5C, buf, 2); break; default: return -EINVAL; - break; } /* write adc power lpf fc */ buf[0] = 0x05; nxt200x_writebytes(state, 0x43, buf, 1); @@ -662,11 +650,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case NXT2004: nxt200x_writebytes(state, 0x4B, buf, 2); break; default: return -EINVAL; - break; } /* write kg1 */ buf[0] = 0x00; nxt200x_writebytes(state, 0x4D, buf, 1); @@ -718,11 +705,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) case VSB_8: buf[0] = 0x00; break; default: return -EINVAL; - break; } nxt200x_writebytes(state, 0x30, buf, 1); /* write agc control reg */ buf[0] = 0x00; @@ -740,11 +726,10 @@ static int nxt200x_setup_frontend_parameters(struct dvb_frontend *fe) nxt200x_writebytes(state, 0x49, buf, 2); nxt200x_writebytes(state, 0x4B, buf, 2); break; default: return -EINVAL; - break; } /* write agc control reg */ buf[0] = 0x04; nxt200x_writebytes(state, 0x41, buf, 1); @@ -1112,11 +1097,10 @@ static int nxt200x_init(struct dvb_frontend* fe) case NXT2004: ret = nxt2004_init(fe); break; default: return -EINVAL; - break; } state->initialised = 1; } return ret; } diff --git a/drivers/media/dvb-frontends/si21xx.c b/drivers/media/dvb-frontends/si21xx.c index a116eff417f2..e31eb2c5cc4c 100644 --- a/drivers/media/dvb-frontends/si21xx.c +++ b/drivers/media/dvb-frontends/si21xx.c @@ -462,14 +462,12 @@ static int si21xx_set_voltage(struct dvb_frontend *fe, enum fe_sec_voltage volt) val = (0x80 | si21_readreg(state, LNB_CTRL_REG_1)); switch (volt) { case SEC_VOLTAGE_18: return si21_writereg(state, LNB_CTRL_REG_1, val | 0x40); - break; case SEC_VOLTAGE_13: return si21_writereg(state, LNB_CTRL_REG_1, (val & ~0x40)); - break; default: return -EINVAL; } } diff --git a/drivers/media/tuners/mt2063.c b/drivers/media/tuners/mt2063.c index 2240d214dfac..d105431a2e2d 100644 --- a/drivers/media/tuners/mt2063.c +++ b/drivers/media/tuners/mt2063.c @@ -1847,11 +1847,10 @@ static int mt2063_init(struct dvb_frontend *fe) def = MT2063B0_defaults; break; default: return -ENODEV; - break; } while (status >= 0 && *def) { u8 reg = *def++; u8 val = *def++; diff --git a/drivers/message/fusion/mptbase.c b/drivers/message/fusion/mptbase.c index 9903e9660a38..3078fac34e51 100644 --- a/drivers/message/fusion/mptbase.c +++ b/drivers/message/fusion/mptbase.c @@ -471,11 +471,10 @@ mpt_turbo_reply(MPT_ADAPTER *ioc, u32 pa) req_idx = pa & 0x0000FFFF; mf = MPT_INDEX_2_MFPTR(ioc, req_idx); mpt_free_msg_frame(ioc, mf); mb(); return; - break; } mr = (MPT_FRAME_HDR *) CAST_U32_TO_PTR(pa); break; case MPI_CONTEXT_REPLY_TYPE_SCSI_TARGET: cb_idx = mpt_get_cb_idx(MPTSTM_DRIVER); diff --git a/drivers/misc/mei/hbm.c b/drivers/misc/mei/hbm.c index a97eb5d47705..686e8b6a4c55 100644 --- a/drivers/misc/mei/hbm.c +++ b/drivers/misc/mei/hbm.c @@ -1375,11 +1375,10 @@ int mei_hbm_dispatch(struct mei_device *dev, struct mei_msg_hdr *hdr) dev->dev_state = MEI_DEV_POWER_DOWN; dev_info(dev->dev, "hbm: stop response: resetting.\n"); /* force the reset */ return -EPROTO; - break; case CLIENT_DISCONNECT_REQ_CMD: dev_dbg(dev->dev, "hbm: disconnect request: message received\n"); disconnect_req = (struct hbm_client_connect_request *)mei_msg; diff --git a/drivers/mtd/mtdchar.c b/drivers/mtd/mtdchar.c index b40f46a43fc6..323035d4f2d0 100644 --- a/drivers/mtd/mtdchar.c +++ b/drivers/mtd/mtdchar.c @@ -879,21 +879,19 @@ static int mtdchar_ioctl(struct file *file, u_int cmd, u_long arg) loff_t offs; if (copy_from_user(&offs, argp, sizeof(loff_t))) return -EFAULT; return mtd_block_isbad(mtd, offs); - break; } case MEMSETBADBLOCK: { loff_t offs; if (copy_from_user(&offs, argp, sizeof(loff_t))) return -EFAULT; return mtd_block_markbad(mtd, offs); - break; } case OTPSELECT: { int mode; diff --git a/drivers/net/can/spi/mcp251xfd/mcp251xfd-core.c b/drivers/net/can/spi/mcp251xfd/mcp251xfd-core.c index c3f49543ff26..9c215f7c5f81 100644 --- a/drivers/net/can/spi/mcp251xfd/mcp251xfd-core.c +++ b/drivers/net/can/spi/mcp251xfd/mcp251xfd-core.c @@ -73,15 +73,15 @@ static const struct can_bittiming_const mcp251xfd_data_bittiming_const = { static const char *__mcp251xfd_get_model_str(enum mcp251xfd_model model) { switch (model) { case MCP251XFD_MODEL_MCP2517FD: - return "MCP2517FD"; break; + return "MCP2517FD"; case MCP251XFD_MODEL_MCP2518FD: - return "MCP2518FD"; break; + return "MCP2518FD"; case MCP251XFD_MODEL_MCP251XFD: - return "MCP251xFD"; break; + return "MCP251xFD"; } return "<unknown>"; } @@ -93,25 +93,25 @@ mcp251xfd_get_model_str(const struct mcp251xfd_priv *priv) static const char *mcp251xfd_get_mode_str(const u8 mode) { switch (mode) { case MCP251XFD_REG_CON_MODE_MIXED: - return "Mixed (CAN FD/CAN 2.0)"; break; + return "Mixed (CAN FD/CAN 2.0)"; case MCP251XFD_REG_CON_MODE_SLEEP: - return "Sleep"; break; + return "Sleep"; case MCP251XFD_REG_CON_MODE_INT_LOOPBACK: - return "Internal Loopback"; break; + return "Internal Loopback"; case MCP251XFD_REG_CON_MODE_LISTENONLY: - return "Listen Only"; break; + return "Listen Only"; case MCP251XFD_REG_CON_MODE_CONFIG: - return "Configuration"; break; + return "Configuration"; case MCP251XFD_REG_CON_MODE_EXT_LOOPBACK: - return "External Loopback"; break; + return "External Loopback"; case MCP251XFD_REG_CON_MODE_CAN2_0: - return "CAN 2.0"; break; + return "CAN 2.0"; case MCP251XFD_REG_CON_MODE_RESTRICTED: - return "Restricted Operation"; break; + return "Restricted Operation"; } return "<unknown>"; } diff --git a/drivers/net/ethernet/aquantia/atlantic/aq_nic.c b/drivers/net/ethernet/aquantia/atlantic/aq_nic.c index 0f865daeb36d..bf5e0e9bd0e2 100644 --- a/drivers/net/ethernet/aquantia/atlantic/aq_nic.c +++ b/drivers/net/ethernet/aquantia/atlantic/aq_nic.c @@ -1161,11 +1161,10 @@ int aq_nic_set_link_ksettings(struct aq_nic_s *self, break; default: err = -1; goto err_exit; - break; } if (!(self->aq_nic_cfg.aq_hw_caps->link_speed_msk & rate)) { err = -1; goto err_exit; } diff --git a/drivers/net/ethernet/cisco/enic/enic_ethtool.c b/drivers/net/ethernet/cisco/enic/enic_ethtool.c index a4dd52bba2c3..1a9803f2073e 100644 --- a/drivers/net/ethernet/cisco/enic/enic_ethtool.c +++ b/drivers/net/ethernet/cisco/enic/enic_ethtool.c @@ -432,11 +432,10 @@ static int enic_grxclsrule(struct enic *enic, struct ethtool_rxnfc *cmd) case IPPROTO_UDP: fsp->flow_type = UDP_V4_FLOW; break; default: return -EINVAL; - break; } fsp->h_u.tcp_ip4_spec.ip4src = flow_get_u32_src(&n->keys); fsp->m_u.tcp_ip4_spec.ip4src = (__u32)~0; diff --git a/drivers/net/ethernet/intel/ixgbe/ixgbe_x540.c b/drivers/net/ethernet/intel/ixgbe/ixgbe_x540.c index de563cfd294d..4b93ba149ec5 100644 --- a/drivers/net/ethernet/intel/ixgbe/ixgbe_x540.c +++ b/drivers/net/ethernet/intel/ixgbe/ixgbe_x540.c @@ -348,11 +348,10 @@ static s32 ixgbe_calc_eeprom_checksum_X540(struct ixgbe_hw *hw) continue; if (ixgbe_read_eerd_generic(hw, pointer, &length)) { hw_dbg(hw, "EEPROM read failed\n"); return IXGBE_ERR_EEPROM; - break; } /* Skip pointer section if length is invalid. */ if (length == 0xFFFF || length == 0 || (pointer + length) >= hw->eeprom.word_size) diff --git a/drivers/net/wan/lmc/lmc_proto.c b/drivers/net/wan/lmc/lmc_proto.c index e8b0b902b424..4e9cc83b615a 100644 --- a/drivers/net/wan/lmc/lmc_proto.c +++ b/drivers/net/wan/lmc/lmc_proto.c @@ -87,21 +87,17 @@ void lmc_proto_close(lmc_softc_t *sc) __be16 lmc_proto_type(lmc_softc_t *sc, struct sk_buff *skb) /*FOLD00*/ { switch(sc->if_type){ case LMC_PPP: return hdlc_type_trans(skb, sc->lmc_device); - break; case LMC_NET: return htons(ETH_P_802_2); - break; case LMC_RAW: /* Packet type for skbuff kind of useless */ return htons(ETH_P_802_2); - break; default: printk(KERN_WARNING "%s: No protocol set for this interface, assuming 802.2 (which is wrong!!)\n", sc->name); return htons(ETH_P_802_2); - break; } } void lmc_proto_netif(lmc_softc_t *sc, struct sk_buff *skb) /*FOLD00*/ { diff --git a/drivers/net/wireless/ath/ath10k/htt_rx.c b/drivers/net/wireless/ath/ath10k/htt_rx.c index 5c1af2021883..9c4e6cf2137a 100644 --- a/drivers/net/wireless/ath/ath10k/htt_rx.c +++ b/drivers/net/wireless/ath/ath10k/htt_rx.c @@ -3876,11 +3876,10 @@ bool ath10k_htt_t2h_msg_handler(struct ath10k *ar, struct sk_buff *skb) atomic_inc(&htt->num_mpdus_ready); return ath10k_htt_rx_proc_rx_frag_ind(htt, &resp->rx_frag_ind, skb); - break; } case HTT_T2H_MSG_TYPE_TEST: break; case HTT_T2H_MSG_TYPE_STATS_CONF: trace_ath10k_htt_stats(ar, skb->data, skb->len); diff --git a/drivers/net/wireless/ath/ath6kl/testmode.c b/drivers/net/wireless/ath/ath6kl/testmode.c index f3906dbe5495..89c7c4e25169 100644 --- a/drivers/net/wireless/ath/ath6kl/testmode.c +++ b/drivers/net/wireless/ath/ath6kl/testmode.c @@ -92,11 +92,10 @@ int ath6kl_tm_cmd(struct wiphy *wiphy, struct wireless_dev *wdev, ath6kl_wmi_test_cmd(ar->wmi, buf, buf_len); return 0; - break; case ATH6KL_TM_CMD_RX_REPORT: default: return -EOPNOTSUPP; } } diff --git a/drivers/net/wireless/ath/ath9k/hw.c b/drivers/net/wireless/ath/ath9k/hw.c index 6609ce122e6e..b66eeb577272 100644 --- a/drivers/net/wireless/ath/ath9k/hw.c +++ b/drivers/net/wireless/ath/ath9k/hw.c @@ -2306,11 +2306,10 @@ void ath9k_hw_beaconinit(struct ath_hw *ah, u32 next_beacon, u32 beacon_period) break; default: ath_dbg(ath9k_hw_common(ah), BEACON, "%s: unsupported opmode: %d\n", __func__, ah->opmode); return; - break; } REG_WRITE(ah, AR_BEACON_PERIOD, beacon_period); REG_WRITE(ah, AR_DMA_BEACON_PERIOD, beacon_period); REG_WRITE(ah, AR_SWBA_PERIOD, beacon_period); diff --git a/drivers/net/wireless/intel/iwlwifi/mvm/mac-ctxt.c b/drivers/net/wireless/intel/iwlwifi/mvm/mac-ctxt.c index cbdebefb854a..8698ca4d30de 100644 --- a/drivers/net/wireless/intel/iwlwifi/mvm/mac-ctxt.c +++ b/drivers/net/wireless/intel/iwlwifi/mvm/mac-ctxt.c @@ -1200,17 +1200,15 @@ static int iwl_mvm_mac_ctx_send(struct iwl_mvm *mvm, struct ieee80211_vif *vif, switch (vif->type) { case NL80211_IFTYPE_STATION: return iwl_mvm_mac_ctxt_cmd_sta(mvm, vif, action, force_assoc_off, bssid_override); - break; case NL80211_IFTYPE_AP: if (!vif->p2p) return iwl_mvm_mac_ctxt_cmd_ap(mvm, vif, action); else return iwl_mvm_mac_ctxt_cmd_go(mvm, vif, action); - break; case NL80211_IFTYPE_MONITOR: return iwl_mvm_mac_ctxt_cmd_listener(mvm, vif, action); case NL80211_IFTYPE_P2P_DEVICE: return iwl_mvm_mac_ctxt_cmd_p2p_device(mvm, vif, action); case NL80211_IFTYPE_ADHOC: diff --git a/drivers/net/wireless/intersil/p54/eeprom.c b/drivers/net/wireless/intersil/p54/eeprom.c index 5bd35c147e19..3ca9d26df174 100644 --- a/drivers/net/wireless/intersil/p54/eeprom.c +++ b/drivers/net/wireless/intersil/p54/eeprom.c @@ -868,11 +868,10 @@ int p54_parse_eeprom(struct ieee80211_hw *dev, void *eeprom, int len) err = -ENOMSG; goto err; } else { goto good_eeprom; } - break; default: break; } crc16 = crc_ccitt(crc16, (u8 *)entry, (entry_len + 1) * 2); diff --git a/drivers/net/wireless/intersil/prism54/oid_mgt.c b/drivers/net/wireless/intersil/prism54/oid_mgt.c index 9fd307ca4b6d..7b251ae90a68 100644 --- a/drivers/net/wireless/intersil/prism54/oid_mgt.c +++ b/drivers/net/wireless/intersil/prism54/oid_mgt.c @@ -785,21 +785,19 @@ mgt_response_to_str(enum oid_num_t n, union oid_res_t *r, char *str) struct obj_buffer *buff = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "size=%u\naddr=0x%X\n", buff->size, buff->addr); } - break; case OID_TYPE_BSS:{ struct obj_bss *bss = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "age=%u\nchannel=%u\n" "capinfo=0x%X\nrates=0x%X\n" "basic_rates=0x%X\n", bss->age, bss->channel, bss->capinfo, bss->rates, bss->basic_rates); } - break; case OID_TYPE_BSSLIST:{ struct obj_bsslist *list = r->ptr; int i, k; k = scnprintf(str, PRIV_STR_SIZE, "nr=%u\n", list->nr); for (i = 0; i < list->nr; i++) @@ -812,53 +810,47 @@ mgt_response_to_str(enum oid_num_t n, union oid_res_t *r, char *str) list->bsslist[i].capinfo, list->bsslist[i].rates, list->bsslist[i].basic_rates); return k; } - break; case OID_TYPE_FREQUENCIES:{ struct obj_frequencies *freq = r->ptr; int i, t; printk("nr : %u\n", freq->nr); t = scnprintf(str, PRIV_STR_SIZE, "nr=%u\n", freq->nr); for (i = 0; i < freq->nr; i++) t += scnprintf(str + t, PRIV_STR_SIZE - t, "mhz[%u]=%u\n", i, freq->mhz[i]); return t; } - break; case OID_TYPE_MLME:{ struct obj_mlme *mlme = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "id=0x%X\nstate=0x%X\ncode=0x%X\n", mlme->id, mlme->state, mlme->code); } - break; case OID_TYPE_MLMEEX:{ struct obj_mlmeex *mlme = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "id=0x%X\nstate=0x%X\n" "code=0x%X\nsize=0x%X\n", mlme->id, mlme->state, mlme->code, mlme->size); } - break; case OID_TYPE_ATTACH:{ struct obj_attachment *attach = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "id=%d\nsize=%d\n", attach->id, attach->size); } - break; case OID_TYPE_SSID:{ struct obj_ssid *ssid = r->ptr; return scnprintf(str, PRIV_STR_SIZE, "length=%u\noctets=%.*s\n", ssid->length, ssid->length, ssid->octets); } - break; case OID_TYPE_KEY:{ struct obj_key *key = r->ptr; int t, i; t = scnprintf(str, PRIV_STR_SIZE, "type=0x%X\nlength=0x%X\nkey=0x", @@ -867,11 +859,10 @@ mgt_response_to_str(enum oid_num_t n, union oid_res_t *r, char *str) t += scnprintf(str + t, PRIV_STR_SIZE - t, "%02X:", key->key[i]); t += scnprintf(str + t, PRIV_STR_SIZE - t, "\n"); return t; } - break; case OID_TYPE_RAW: case OID_TYPE_ADDR:{ unsigned char *buff = r->ptr; int t, i; t = scnprintf(str, PRIV_STR_SIZE, "hex data="); @@ -879,11 +870,10 @@ mgt_response_to_str(enum oid_num_t n, union oid_res_t *r, char *str) t += scnprintf(str + t, PRIV_STR_SIZE - t, "%02X:", buff[i]); t += scnprintf(str + t, PRIV_STR_SIZE - t, "\n"); return t; } - break; default: BUG(); } return 0; } diff --git a/drivers/net/wireless/realtek/rtlwifi/rtl8188ee/hw.c b/drivers/net/wireless/realtek/rtlwifi/rtl8188ee/hw.c index 63f9ea21962f..bd9160b166c5 100644 --- a/drivers/net/wireless/realtek/rtlwifi/rtl8188ee/hw.c +++ b/drivers/net/wireless/realtek/rtlwifi/rtl8188ee/hw.c @@ -1224,11 +1224,10 @@ static int _rtl88ee_set_media_status(struct ieee80211_hw *hw, "Set Network type to AP!\n"); break; default: pr_err("Network type %d not support!\n", type); return 1; - break; } /* MSR_INFRA == Link in infrastructure network; * MSR_ADHOC == Link in ad hoc network; * Therefore, check link state is necessary. diff --git a/drivers/net/wireless/realtek/rtlwifi/rtl8723ae/hw.c b/drivers/net/wireless/realtek/rtlwifi/rtl8723ae/hw.c index a36dc6e726d2..f8a1de6e9849 100644 --- a/drivers/net/wireless/realtek/rtlwifi/rtl8723ae/hw.c +++ b/drivers/net/wireless/realtek/rtlwifi/rtl8723ae/hw.c @@ -1130,11 +1130,10 @@ static int _rtl8723e_set_media_status(struct ieee80211_hw *hw, "Set Network type to AP!\n"); break; default: pr_err("Network type %d not support!\n", type); return 1; - break; } /* MSR_INFRA == Link in infrastructure network; * MSR_ADHOC == Link in ad hoc network; * Therefore, check link state is necessary. diff --git a/drivers/net/wireless/realtek/rtlwifi/rtl8821ae/phy.c b/drivers/net/wireless/realtek/rtlwifi/rtl8821ae/phy.c index f41a7643b9c4..225b8cd44f23 100644 --- a/drivers/net/wireless/realtek/rtlwifi/rtl8821ae/phy.c +++ b/drivers/net/wireless/realtek/rtlwifi/rtl8821ae/phy.c @@ -2083,16 +2083,14 @@ bool rtl8812ae_phy_config_rf_with_headerfile(struct ieee80211_hw *hw, switch (rfpath) { case RF90_PATH_A: return __rtl8821ae_phy_config_with_headerfile(hw, radioa_array_table_a, radioa_arraylen_a, _rtl8821ae_config_rf_radio_a); - break; case RF90_PATH_B: return __rtl8821ae_phy_config_with_headerfile(hw, radioa_array_table_b, radioa_arraylen_b, _rtl8821ae_config_rf_radio_b); - break; case RF90_PATH_C: case RF90_PATH_D: pr_err("switch case %#x not processed\n", rfpath); break; } @@ -2114,11 +2112,10 @@ bool rtl8821ae_phy_config_rf_with_headerfile(struct ieee80211_hw *hw, switch (rfpath) { case RF90_PATH_A: return __rtl8821ae_phy_config_with_headerfile(hw, radioa_array_table, radioa_arraylen, _rtl8821ae_config_rf_radio_a); - break; case RF90_PATH_B: case RF90_PATH_C: case RF90_PATH_D: pr_err("switch case %#x not processed\n", rfpath); diff --git a/drivers/nfc/st21nfca/core.c b/drivers/nfc/st21nfca/core.c index 2ce17932a073..6ca0d2f56b18 100644 --- a/drivers/nfc/st21nfca/core.c +++ b/drivers/nfc/st21nfca/core.c @@ -792,11 +792,10 @@ static int st21nfca_hci_im_transceive(struct nfc_hci_dev *hdev, return nfc_hci_send_cmd_async(hdev, target->hci_reader_gate, ST21NFCA_WR_XCHG_DATA, skb->data, skb->len, st21nfca_hci_data_exchange_cb, info); - break; default: return 1; } } diff --git a/drivers/nfc/trf7970a.c b/drivers/nfc/trf7970a.c index 3bd97c73f983..c70f62fe321e 100644 --- a/drivers/nfc/trf7970a.c +++ b/drivers/nfc/trf7970a.c @@ -1380,11 +1380,10 @@ static int trf7970a_is_iso15693_write_or_lock(u8 cmd) case ISO15693_CMD_WRITE_AFI: case ISO15693_CMD_LOCK_AFI: case ISO15693_CMD_WRITE_DSFID: case ISO15693_CMD_LOCK_DSFID: return 1; - break; default: return 0; } } diff --git a/drivers/nvdimm/claim.c b/drivers/nvdimm/claim.c index 5a7c80053c62..2f250874b1a4 100644 --- a/drivers/nvdimm/claim.c +++ b/drivers/nvdimm/claim.c @@ -200,11 +200,10 @@ ssize_t nd_namespace_store(struct device *dev, } break; default: len = -EBUSY; goto out_attach; - break; } if (__nvdimm_namespace_capacity(ndns) < SZ_16M) { dev_dbg(dev, "%s too small to host\n", name); len = -ENXIO; diff --git a/drivers/pci/controller/pci-v3-semi.c b/drivers/pci/controller/pci-v3-semi.c index 1f54334f09f7..154a5398633c 100644 --- a/drivers/pci/controller/pci-v3-semi.c +++ b/drivers/pci/controller/pci-v3-semi.c @@ -656,11 +656,10 @@ static int v3_get_dma_range_config(struct v3_pci *v3, val |= V3_LB_BASE_ADR_SIZE_2GB; break; default: dev_err(v3->dev, "illegal dma memory chunk size\n"); return -EINVAL; - break; } val |= V3_PCI_MAP_M_REG_EN | V3_PCI_MAP_M_ENABLE; *pci_map = val; dev_dbg(dev, diff --git a/drivers/pinctrl/samsung/pinctrl-s3c24xx.c b/drivers/pinctrl/samsung/pinctrl-s3c24xx.c index 5e24838a582f..2223ead5bd72 100644 --- a/drivers/pinctrl/samsung/pinctrl-s3c24xx.c +++ b/drivers/pinctrl/samsung/pinctrl-s3c24xx.c @@ -106,23 +106,18 @@ struct s3c24xx_eint_domain_data { static int s3c24xx_eint_get_trigger(unsigned int type) { switch (type) { case IRQ_TYPE_EDGE_RISING: return EINT_EDGE_RISING; - break; case IRQ_TYPE_EDGE_FALLING: return EINT_EDGE_FALLING; - break; case IRQ_TYPE_EDGE_BOTH: return EINT_EDGE_BOTH; - break; case IRQ_TYPE_LEVEL_HIGH: return EINT_LEVEL_HIGH; - break; case IRQ_TYPE_LEVEL_LOW: return EINT_LEVEL_LOW; - break; default: return -EINVAL; } } diff --git a/drivers/platform/x86/acer-wmi.c b/drivers/platform/x86/acer-wmi.c index 49f4b73be513..1c2084c74a57 100644 --- a/drivers/platform/x86/acer-wmi.c +++ b/drivers/platform/x86/acer-wmi.c @@ -790,11 +790,10 @@ static acpi_status AMW0_set_u32(u32 value, u32 cap) if (value > max_brightness) return AE_BAD_PARAMETER; switch (quirks->brightness) { default: return ec_write(0x83, value); - break; } default: return AE_ERROR; } diff --git a/drivers/platform/x86/sony-laptop.c b/drivers/platform/x86/sony-laptop.c index e5a1b5533408..704813374922 100644 --- a/drivers/platform/x86/sony-laptop.c +++ b/drivers/platform/x86/sony-laptop.c @@ -2465,26 +2465,23 @@ static int __sony_nc_gfx_switch_status_get(void) case 0x0146: /* 1: discrete GFX (speed) * 0: integrated GFX (stamina) */ return result & 0x1 ? SPEED : STAMINA; - break; case 0x015B: /* 0: discrete GFX (speed) * 1: integrated GFX (stamina) */ return result & 0x1 ? STAMINA : SPEED; - break; case 0x0128: /* it's a more elaborated bitmask, for now: * 2: integrated GFX (stamina) * 0: discrete GFX (speed) */ dprintk("GFX Status: 0x%x\n", result); return result & 0x80 ? AUTO : result & 0x02 ? STAMINA : SPEED; - break; } return -EINVAL; } static ssize_t sony_nc_gfx_switch_status_show(struct device *dev, diff --git a/drivers/platform/x86/wmi.c b/drivers/platform/x86/wmi.c index d88f388a3450..44e802f9f1b4 100644 --- a/drivers/platform/x86/wmi.c +++ b/drivers/platform/x86/wmi.c @@ -1258,17 +1258,14 @@ acpi_wmi_ec_space_handler(u32 function, acpi_physical_address address, } switch (result) { case -EINVAL: return AE_BAD_PARAMETER; - break; case -ENODEV: return AE_NOT_FOUND; - break; case -ETIME: return AE_TIME; - break; default: return AE_OK; } } diff --git a/drivers/power/supply/wm831x_power.c b/drivers/power/supply/wm831x_power.c index 18b33f14dfee..4cd2dd870039 100644 --- a/drivers/power/supply/wm831x_power.c +++ b/drivers/power/supply/wm831x_power.c @@ -666,11 +666,10 @@ static int wm831x_power_probe(struct platform_device *pdev) default: dev_err(&pdev->dev, "Failed to find USB phy: %d\n", ret); fallthrough; case -EPROBE_DEFER: goto err_bat_irq; - break; } return ret; err_bat_irq: diff --git a/drivers/scsi/aic94xx/aic94xx_task.c b/drivers/scsi/aic94xx/aic94xx_task.c index f923ed019d4a..ed034192b3c3 100644 --- a/drivers/scsi/aic94xx/aic94xx_task.c +++ b/drivers/scsi/aic94xx/aic94xx_task.c @@ -267,11 +267,10 @@ static void asd_task_tasklet_complete(struct asd_ascb *ascb, ts->stat = SAS_NAK_R_ERR; break; case TA_I_T_NEXUS_LOSS: opcode = dl->status_block[0]; goto Again; - break; case TF_INV_CONN_HANDLE: ts->resp = SAS_TASK_UNDELIVERED; ts->stat = SAS_DEVICE_UNKNOWN; break; case TF_REQUESTED_N_PENDING: diff --git a/drivers/scsi/be2iscsi/be_mgmt.c b/drivers/scsi/be2iscsi/be_mgmt.c index 96d6e384b2b2..0d4928567265 100644 --- a/drivers/scsi/be2iscsi/be_mgmt.c +++ b/drivers/scsi/be2iscsi/be_mgmt.c @@ -1242,22 +1242,18 @@ beiscsi_adap_family_disp(struct device *dev, struct device_attribute *attr, case BE_DEVICE_ID1: case OC_DEVICE_ID1: case OC_DEVICE_ID2: return snprintf(buf, PAGE_SIZE, "Obsolete/Unsupported BE2 Adapter Family\n"); - break; case BE_DEVICE_ID2: case OC_DEVICE_ID3: return snprintf(buf, PAGE_SIZE, "BE3-R Adapter Family\n"); - break; case OC_SKH_ID1: return snprintf(buf, PAGE_SIZE, "Skyhawk-R Adapter Family\n"); - break; default: return snprintf(buf, PAGE_SIZE, "Unknown Adapter Family: 0x%x\n", dev_id); - break; } } /** * beiscsi_phys_port()- Display Physical Port Identifier diff --git a/drivers/scsi/bnx2fc/bnx2fc_hwi.c b/drivers/scsi/bnx2fc/bnx2fc_hwi.c index 08992095ce7a..b37b0a9ec12d 100644 --- a/drivers/scsi/bnx2fc/bnx2fc_hwi.c +++ b/drivers/scsi/bnx2fc/bnx2fc_hwi.c @@ -768,11 +768,10 @@ static void bnx2fc_process_unsol_compl(struct bnx2fc_rport *tgt, u16 wqe) if (rc) goto skip_rec; } else printk(KERN_ERR PFX "SRR in progress\n"); goto ret_err_rqe; - break; default: break; } skip_rec: diff --git a/drivers/scsi/fcoe/fcoe.c b/drivers/scsi/fcoe/fcoe.c index 0f9274960dc6..a4be6f439c47 100644 --- a/drivers/scsi/fcoe/fcoe.c +++ b/drivers/scsi/fcoe/fcoe.c @@ -1892,11 +1892,10 @@ static int fcoe_device_notification(struct notifier_block *notifier, fcoe_interface_remove(fcoe); fcoe_interface_cleanup(fcoe); mutex_unlock(&fcoe_config_mutex); fcoe_ctlr_device_delete(fcoe_ctlr_to_ctlr_dev(ctlr)); goto out; - break; case NETDEV_FEAT_CHANGE: fcoe_netdev_features_change(lport, netdev); break; default: FCOE_NETDEV_DBG(netdev, "Unknown event %ld " diff --git a/drivers/scsi/hpsa.c b/drivers/scsi/hpsa.c index 83ce4f11a589..45136e3a4efc 100644 --- a/drivers/scsi/hpsa.c +++ b/drivers/scsi/hpsa.c @@ -7440,11 +7440,10 @@ static int find_PCI_BAR_index(struct pci_dev *pdev, unsigned long pci_bar_addr) break; default: /* reserved in PCI 2.2 */ dev_warn(&pdev->dev, "base address is invalid\n"); return -1; - break; } } if (offset == pci_bar_addr - PCI_BASE_ADDRESS_0) return i + 1; } diff --git a/drivers/scsi/hptiop.c b/drivers/scsi/hptiop.c index 6a2561f26e38..db4c7a7ff4dd 100644 --- a/drivers/scsi/hptiop.c +++ b/drivers/scsi/hptiop.c @@ -756,11 +756,10 @@ static void hptiop_finish_scsi_req(struct hptiop_hba *hba, u32 tag, scsi_set_resid(scp, scsi_bufflen(scp) - le32_to_cpu(req->dataxfer_length)); scp->result = SAM_STAT_CHECK_CONDITION; memcpy(scp->sense_buffer, &req->sg_list, SCSI_SENSE_BUFFERSIZE); goto skip_resid; - break; default: scp->result = DRIVER_INVALID << 24 | DID_ABORT << 16; break; } diff --git a/drivers/scsi/ipr.c b/drivers/scsi/ipr.c index b0aa58d117cc..e451102b9a29 100644 --- a/drivers/scsi/ipr.c +++ b/drivers/scsi/ipr.c @@ -9485,11 +9485,10 @@ static pci_ers_result_t ipr_pci_error_detected(struct pci_dev *pdev, ipr_pci_frozen(pdev); return PCI_ERS_RESULT_CAN_RECOVER; case pci_channel_io_perm_failure: ipr_pci_perm_failure(pdev); return PCI_ERS_RESULT_DISCONNECT; - break; default: break; } return PCI_ERS_RESULT_NEED_RESET; } diff --git a/drivers/scsi/isci/phy.c b/drivers/scsi/isci/phy.c index 7041e2e3ab48..1b87d9080ebe 100644 --- a/drivers/scsi/isci/phy.c +++ b/drivers/scsi/isci/phy.c @@ -751,11 +751,10 @@ enum sci_status sci_phy_event_handler(struct isci_phy *iphy, u32 event_code) sci_change_state(&iphy->sm, SCI_PHY_STARTING); break; default: phy_event_warn(iphy, state, event_code); return SCI_FAILURE; - break; } return SCI_SUCCESS; case SCI_PHY_SUB_AWAIT_IAF_UF: switch (scu_get_event_code(event_code)) { case SCU_EVENT_SAS_PHY_DETECTED: @@ -956,11 +955,10 @@ enum sci_status sci_phy_event_handler(struct isci_phy *iphy, u32 event_code) sci_change_state(&iphy->sm, SCI_PHY_STARTING); break; default: phy_event_warn(iphy, state, event_code); return SCI_FAILURE_INVALID_STATE; - break; } return SCI_SUCCESS; default: dev_dbg(sciphy_to_dev(iphy), "%s: in wrong state: %s\n", __func__, phy_state_name(state)); diff --git a/drivers/scsi/lpfc/lpfc_debugfs.c b/drivers/scsi/lpfc/lpfc_debugfs.c index c9a327b13e5c..325081ac6553 100644 --- a/drivers/scsi/lpfc/lpfc_debugfs.c +++ b/drivers/scsi/lpfc/lpfc_debugfs.c @@ -3339,11 +3339,10 @@ lpfc_idiag_pcicfg_read(struct file *file, char __user *buf, size_t nbytes, len += scnprintf(pbuffer+len, LPFC_PCI_CFG_SIZE-len, "%03x: %08x\n", where, u32val); break; case LPFC_PCI_CFG_BROWSE: /* browse all */ goto pcicfg_browse; - break; default: /* illegal count */ len = 0; break; } @@ -4379,11 +4378,11 @@ lpfc_idiag_queacc_write(struct file *file, const char __user *buf, goto pass_check; } } } goto error_out; - break; + case LPFC_IDIAG_CQ: /* MBX complete queue */ if (phba->sli4_hba.mbx_cq && phba->sli4_hba.mbx_cq->queue_id == queid) { /* Sanity check */ @@ -4431,11 +4430,11 @@ lpfc_idiag_queacc_write(struct file *file, const char __user *buf, goto pass_check; } } } goto error_out; - break; + case LPFC_IDIAG_MQ: /* MBX work queue */ if (phba->sli4_hba.mbx_wq && phba->sli4_hba.mbx_wq->queue_id == queid) { /* Sanity check */ @@ -4445,11 +4444,11 @@ lpfc_idiag_queacc_write(struct file *file, const char __user *buf, goto error_out; idiag.ptr_private = phba->sli4_hba.mbx_wq; goto pass_check; } goto error_out; - break; + case LPFC_IDIAG_WQ: /* ELS work queue */ if (phba->sli4_hba.els_wq && phba->sli4_hba.els_wq->queue_id == queid) { /* Sanity check */ @@ -4485,13 +4484,12 @@ lpfc_idiag_queacc_write(struct file *file, const char __user *buf, idiag.ptr_private = qp; goto pass_check; } } } - goto error_out; - break; + case LPFC_IDIAG_RQ: /* HDR queue */ if (phba->sli4_hba.hdr_rq && phba->sli4_hba.hdr_rq->queue_id == queid) { /* Sanity check */ @@ -4512,14 +4510,12 @@ lpfc_idiag_queacc_write(struct file *file, const char __user *buf, goto error_out; idiag.ptr_private = phba->sli4_hba.dat_rq; goto pass_check; } goto error_out; - break; default: goto error_out; - break; } pass_check: if (idiag.cmd.opcode == LPFC_IDIAG_CMD_QUEACC_RD) { diff --git a/drivers/scsi/lpfc/lpfc_init.c b/drivers/scsi/lpfc/lpfc_init.c index ca25e54bb782..b6090357e8a5 100644 --- a/drivers/scsi/lpfc/lpfc_init.c +++ b/drivers/scsi/lpfc/lpfc_init.c @@ -7194,11 +7194,10 @@ lpfc_init_api_table_setup(struct lpfc_hba *phba, uint8_t dev_grp) default: lpfc_printf_log(phba, KERN_ERR, LOG_TRACE_EVENT, "1431 Invalid HBA PCI-device group: 0x%x\n", dev_grp); return -ENODEV; - break; } return 0; } /** diff --git a/drivers/scsi/lpfc/lpfc_scsi.c b/drivers/scsi/lpfc/lpfc_scsi.c index 983eeb0e3d07..c3b02dab6e5c 100644 --- a/drivers/scsi/lpfc/lpfc_scsi.c +++ b/drivers/scsi/lpfc/lpfc_scsi.c @@ -4282,11 +4282,10 @@ lpfc_scsi_api_table_setup(struct lpfc_hba *phba, uint8_t dev_grp) default: lpfc_printf_log(phba, KERN_ERR, LOG_TRACE_EVENT, "1418 Invalid HBA PCI-device group: 0x%x\n", dev_grp); return -ENODEV; - break; } phba->lpfc_rampdown_queue_depth = lpfc_rampdown_queue_depth; phba->lpfc_scsi_cmd_iocb_cmpl = lpfc_scsi_cmd_iocb_cmpl; return 0; } diff --git a/drivers/scsi/lpfc/lpfc_sli.c b/drivers/scsi/lpfc/lpfc_sli.c index e158cd77d387..0f18f1ba8a28 100644 --- a/drivers/scsi/lpfc/lpfc_sli.c +++ b/drivers/scsi/lpfc/lpfc_sli.c @@ -9187,11 +9187,10 @@ lpfc_mbox_api_table_setup(struct lpfc_hba *phba, uint8_t dev_grp) default: lpfc_printf_log(phba, KERN_ERR, LOG_TRACE_EVENT, "1420 Invalid HBA PCI-device group: 0x%x\n", dev_grp); return -ENODEV; - break; } return 0; } /** @@ -10070,11 +10069,10 @@ lpfc_sli4_iocb2wqe(struct lpfc_hba *phba, struct lpfc_iocbq *iocbq, default: lpfc_printf_log(phba, KERN_ERR, LOG_TRACE_EVENT, "2014 Invalid command 0x%x\n", iocbq->iocb.ulpCommand); return IOCB_ERROR; - break; } if (iocbq->iocb_flag & LPFC_IO_DIF_PASS) bf_set(wqe_dif, &wqe->generic.wqe_com, LPFC_WQE_DIF_PASSTHRU); else if (iocbq->iocb_flag & LPFC_IO_DIF_STRIP) @@ -10232,11 +10230,10 @@ lpfc_sli_api_table_setup(struct lpfc_hba *phba, uint8_t dev_grp) default: lpfc_printf_log(phba, KERN_ERR, LOG_TRACE_EVENT, "1419 Invalid HBA PCI-device group: 0x%x\n", dev_grp); return -ENODEV; - break; } phba->lpfc_get_iocb_from_iocbq = lpfc_get_iocb_from_iocbq; return 0; } diff --git a/drivers/scsi/mvumi.c b/drivers/scsi/mvumi.c index 0354898d7cac..2f7a52bd653a 100644 --- a/drivers/scsi/mvumi.c +++ b/drivers/scsi/mvumi.c @@ -2294,11 +2294,10 @@ static int mvumi_cfg_hw_reg(struct mvumi_hba *mhba) regs->int_drbl_int_mask = 0x3FFFFFFF; regs->int_mu = regs->int_dl_cpu2pciea | regs->int_comaout; break; default: return -1; - break; } return 0; } diff --git a/drivers/scsi/pcmcia/nsp_cs.c b/drivers/scsi/pcmcia/nsp_cs.c index bc5a623519e7..bb3b3884f968 100644 --- a/drivers/scsi/pcmcia/nsp_cs.c +++ b/drivers/scsi/pcmcia/nsp_cs.c @@ -1100,12 +1100,10 @@ static irqreturn_t nspintr(int irq, void *dev_id) nsp_index_write(base, SCSIBUSCTRL, SCSI_ATN); udelay(1); nsp_index_write(base, SCSIBUSCTRL, SCSI_ATN | AUTODIRECTION | ACKENB); return IRQ_HANDLED; - break; - case PH_RESELECT: //nsp_dbg(NSP_DEBUG_INTR, "phase reselect"); // *sync_neg = SYNC_NOT_YET; if ((phase & BUSMON_PHASE_MASK) != BUSPHASE_MESSAGE_IN) { diff --git a/drivers/scsi/qla2xxx/qla_mbx.c b/drivers/scsi/qla2xxx/qla_mbx.c index 07afd0d8a8f3..40af7f1524ce 100644 --- a/drivers/scsi/qla2xxx/qla_mbx.c +++ b/drivers/scsi/qla2xxx/qla_mbx.c @@ -4028,11 +4028,10 @@ qla24xx_report_id_acquisition(scsi_qla_host_t *vha, /* if our portname is higher then initiate N2N login */ set_bit(N2N_LOGIN_NEEDED, &vha->dpc_flags); return; - break; case TOPO_FL: ha->current_topology = ISP_CFG_FL; break; case TOPO_F: ha->current_topology = ISP_CFG_F; diff --git a/drivers/scsi/st.c b/drivers/scsi/st.c index e2e5356a997d..43f7624508a9 100644 --- a/drivers/scsi/st.c +++ b/drivers/scsi/st.c @@ -2844,11 +2844,10 @@ static int st_int_ioctl(struct scsi_tape *STp, unsigned int cmd_in, unsigned lon fileno = blkno = at_sm = 0; break; case MTNOP: DEBC_printk(STp, "No op on tape.\n"); return 0; /* Should do something ? */ - break; case MTRETEN: cmd[0] = START_STOP; if (STp->immediate) { cmd[1] = 1; /* Don't wait for completion */ timeout = STp->device->request_queue->rq_timeout; diff --git a/drivers/scsi/sym53c8xx_2/sym_hipd.c b/drivers/scsi/sym53c8xx_2/sym_hipd.c index a9fe092a4906..255a2d48d421 100644 --- a/drivers/scsi/sym53c8xx_2/sym_hipd.c +++ b/drivers/scsi/sym53c8xx_2/sym_hipd.c @@ -4594,11 +4594,10 @@ static void sym_int_sir(struct sym_hcb *np) sym_print_addr(cp->cmd, "M_REJECT received (%x:%x).\n", scr_to_cpu(np->lastmsg), np->msgout[0]); } goto out_clrack; - break; default: goto out_reject; } break; /* diff --git a/drivers/staging/media/atomisp/pci/sh_css.c b/drivers/staging/media/atomisp/pci/sh_css.c index ddee04c8248d..eab7a048ca7a 100644 --- a/drivers/staging/media/atomisp/pci/sh_css.c +++ b/drivers/staging/media/atomisp/pci/sh_css.c @@ -6284,11 +6284,10 @@ allocate_delay_frames(struct ia_css_pipe *pipe) { case IA_CSS_PIPE_ID_CAPTURE: { struct ia_css_capture_settings *mycs_capture = &pipe->pipe_settings.capture; (void)mycs_capture; return err; } - break; case IA_CSS_PIPE_ID_VIDEO: { struct ia_css_video_settings *mycs_video = &pipe->pipe_settings.video; ref_info = mycs_video->video_binary.internal_frame_info; /*The ref frame expects diff --git a/drivers/staging/rts5208/rtsx_scsi.c b/drivers/staging/rts5208/rtsx_scsi.c index 1deb74112ad4..672248be7bf3 100644 --- a/drivers/staging/rts5208/rtsx_scsi.c +++ b/drivers/staging/rts5208/rtsx_scsi.c @@ -569,12 +569,10 @@ static int start_stop_unit(struct scsi_cmnd *srb, struct rtsx_chip *chip) case LOAD_MEDIUM: if (check_card_ready(chip, lun)) return TRANSPORT_GOOD; set_sense_type(chip, lun, SENSE_TYPE_MEDIA_NOT_PRESENT); return TRANSPORT_FAILED; - - break; } return TRANSPORT_ERROR; } diff --git a/drivers/staging/vme/devices/vme_user.c b/drivers/staging/vme/devices/vme_user.c index fd0ea4dbcb91..7c7d3858e6ca 100644 --- a/drivers/staging/vme/devices/vme_user.c +++ b/drivers/staging/vme/devices/vme_user.c @@ -355,12 +355,10 @@ static int vme_user_ioctl(struct inode *inode, struct file *file, * to userspace as they are */ return vme_master_set(image[minor].resource, master.enable, master.vme_addr, master.size, master.aspace, master.cycle, master.dwidth); - - break; } break; case SLAVE_MINOR: switch (cmd) { case VME_GET_SLAVE: @@ -396,12 +394,10 @@ static int vme_user_ioctl(struct inode *inode, struct file *file, */ return vme_slave_set(image[minor].resource, slave.enable, slave.vme_addr, slave.size, image[minor].pci_buf, slave.aspace, slave.cycle); - - break; } break; } return -EINVAL; diff --git a/drivers/tty/nozomi.c b/drivers/tty/nozomi.c index d42b854cb7df..861e95043191 100644 --- a/drivers/tty/nozomi.c +++ b/drivers/tty/nozomi.c @@ -412,15 +412,13 @@ static void read_mem32(u32 *buf, const void __iomem *mem_addr_start, switch (size_bytes) { case 2: /* 2 bytes */ buf16 = (u16 *) buf; *buf16 = __le16_to_cpu(readw(ptr)); goto out; - break; case 4: /* 4 bytes */ *(buf) = __le32_to_cpu(readl(ptr)); goto out; - break; } while (i < size_bytes) { if (size_bytes - i == 2) { /* Handle 2 bytes in the end */ @@ -458,19 +456,18 @@ static u32 write_mem32(void __iomem *mem_addr_start, const u32 *buf, switch (size_bytes) { case 2: /* 2 bytes */ buf16 = (const u16 *)buf; writew(__cpu_to_le16(*buf16), ptr); return 2; - break; case 1: /* * also needs to write 4 bytes in this case * so falling through.. */ + fallthrough; case 4: /* 4 bytes */ writel(__cpu_to_le32(*buf), ptr); return 4; - break; } while (i < size_bytes) { if (size_bytes - i == 2) { /* 2 bytes */ diff --git a/drivers/tty/serial/imx.c b/drivers/tty/serial/imx.c index 1731d9728865..09703079db7b 100644 --- a/drivers/tty/serial/imx.c +++ b/drivers/tty/serial/imx.c @@ -318,31 +318,26 @@ static void imx_uart_writel(struct imx_port *sport, u32 val, u32 offset) static u32 imx_uart_readl(struct imx_port *sport, u32 offset) { switch (offset) { case UCR1: return sport->ucr1; - break; case UCR2: /* * UCR2_SRST is the only bit in the cached registers that might * differ from the value that was last written. As it only * automatically becomes one after being cleared, reread * conditionally. */ if (!(sport->ucr2 & UCR2_SRST)) sport->ucr2 = readl(sport->port.membase + offset); return sport->ucr2; - break; case UCR3: return sport->ucr3; - break; case UCR4: return sport->ucr4; - break; case UFCR: return sport->ufcr; - break; default: return readl(sport->port.membase + offset); } } diff --git a/drivers/usb/gadget/function/f_hid.c b/drivers/usb/gadget/function/f_hid.c index 1125f4715830..5204769834d1 100644 --- a/drivers/usb/gadget/function/f_hid.c +++ b/drivers/usb/gadget/function/f_hid.c @@ -509,27 +509,23 @@ static int hidg_setup(struct usb_function *f, VDBG(cdev, "get_report\n"); /* send an empty report */ length = min_t(unsigned, length, hidg->report_length); memset(req->buf, 0x0, length); - goto respond; - break; case ((USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE) << 8 | HID_REQ_GET_PROTOCOL): VDBG(cdev, "get_protocol\n"); length = min_t(unsigned int, length, 1); ((u8 *) req->buf)[0] = hidg->protocol; goto respond; - break; case ((USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_INTERFACE) << 8 | HID_REQ_SET_REPORT): VDBG(cdev, "set_report | wLength=%d\n", ctrl->wLength); goto stall; - break; case ((USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_INTERFACE) << 8 | HID_REQ_SET_PROTOCOL): VDBG(cdev, "set_protocol\n"); if (value > HID_REPORT_PROTOCOL) @@ -542,11 +538,10 @@ static int hidg_setup(struct usb_function *f, if (hidg->bInterfaceSubClass == USB_INTERFACE_SUBCLASS_BOOT) { hidg->protocol = value; goto respond; } goto stall; - break; case ((USB_DIR_IN | USB_TYPE_STANDARD | USB_RECIP_INTERFACE) << 8 | USB_REQ_GET_DESCRIPTOR): switch (value >> 8) { case HID_DT_HID: @@ -560,33 +555,29 @@ static int hidg_setup(struct usb_function *f, length = min_t(unsigned short, length, hidg_desc_copy.bLength); memcpy(req->buf, &hidg_desc_copy, length); goto respond; - break; } case HID_DT_REPORT: VDBG(cdev, "USB_REQ_GET_DESCRIPTOR: REPORT\n"); length = min_t(unsigned short, length, hidg->report_desc_length); memcpy(req->buf, hidg->report_desc, length); goto respond; - break; default: VDBG(cdev, "Unknown descriptor request 0x%x\n", value >> 8); goto stall; - break; } break; default: VDBG(cdev, "Unknown request 0x%x\n", ctrl->bRequest); goto stall; - break; } stall: return -EOPNOTSUPP; diff --git a/drivers/usb/host/xhci-mem.c b/drivers/usb/host/xhci-mem.c index fe405cd38dbc..b46ef45c4d25 100644 --- a/drivers/usb/host/xhci-mem.c +++ b/drivers/usb/host/xhci-mem.c @@ -1142,11 +1142,10 @@ int xhci_setup_addressable_virt_dev(struct xhci_hcd *xhci, struct usb_device *ud max_packets = MAX_PACKET(8); break; case USB_SPEED_WIRELESS: xhci_dbg(xhci, "FIXME xHCI doesn't support wireless speeds\n"); return -EINVAL; - break; default: /* Speed was set earlier, this shouldn't happen. */ return -EINVAL; } /* Find the root hub port this device is under */ diff --git a/drivers/usb/misc/iowarrior.c b/drivers/usb/misc/iowarrior.c index 70ec29681526..efbd317f2f25 100644 --- a/drivers/usb/misc/iowarrior.c +++ b/drivers/usb/misc/iowarrior.c @@ -382,11 +382,10 @@ static ssize_t iowarrior_write(struct file *file, goto exit; } retval = usb_set_report(dev->interface, 2, 0, buf, count); kfree(buf); goto exit; - break; case USB_DEVICE_ID_CODEMERCS_IOW56: case USB_DEVICE_ID_CODEMERCS_IOW56AM: case USB_DEVICE_ID_CODEMERCS_IOW28: case USB_DEVICE_ID_CODEMERCS_IOW28L: case USB_DEVICE_ID_CODEMERCS_IOW100: @@ -452,18 +451,16 @@ static ssize_t iowarrior_write(struct file *file, } /* submit was ok */ retval = count; usb_free_urb(int_out_urb); goto exit; - break; default: /* what do we have here ? An unsupported Product-ID ? */ dev_err(&dev->interface->dev, "%s - not supported for product=0x%x\n", __func__, dev->product_id); retval = -EFAULT; goto exit; - break; } error: usb_free_coherent(dev->udev, dev->report_size, buf, int_out_urb->transfer_dma); error_no_buffer: diff --git a/drivers/usb/serial/iuu_phoenix.c b/drivers/usb/serial/iuu_phoenix.c index b4ba79123d9d..f1201d4de297 100644 --- a/drivers/usb/serial/iuu_phoenix.c +++ b/drivers/usb/serial/iuu_phoenix.c @@ -848,11 +848,10 @@ static int iuu_uart_baud(struct usb_serial_port *port, u32 baud_base, dataout[DataCount++] = 0x04; break; default: kfree(dataout); return IUU_INVALID_PARAMETER; - break; } switch (parity & 0xF0) { case IUU_ONE_STOP_BIT: dataout[DataCount - 1] |= IUU_ONE_STOP_BIT; @@ -862,11 +861,10 @@ static int iuu_uart_baud(struct usb_serial_port *port, u32 baud_base, dataout[DataCount - 1] |= IUU_TWO_STOP_BITS; break; default: kfree(dataout); return IUU_INVALID_PARAMETER; - break; } status = bulk_immediate(port, dataout, DataCount); if (status != IUU_OPERATION_OK) dev_dbg(&port->dev, "%s - uart_off error\n", __func__); diff --git a/drivers/usb/storage/freecom.c b/drivers/usb/storage/freecom.c index 3d5f7d0ff0f1..2b098b55c4cb 100644 --- a/drivers/usb/storage/freecom.c +++ b/drivers/usb/storage/freecom.c @@ -429,11 +429,10 @@ static int freecom_transport(struct scsi_cmnd *srb, struct us_data *us) /* should never hit here -- filtered in usb.c */ usb_stor_dbg(us, "freecom unimplemented direction: %d\n", us->srb->sc_data_direction); /* Return fail, SCSI seems to handle this better. */ return USB_STOR_TRANSPORT_FAILED; - break; } return USB_STOR_TRANSPORT_GOOD; } diff --git a/drivers/vme/bridges/vme_tsi148.c b/drivers/vme/bridges/vme_tsi148.c index 50ae26977a02..1227ea937059 100644 --- a/drivers/vme/bridges/vme_tsi148.c +++ b/drivers/vme/bridges/vme_tsi148.c @@ -504,11 +504,10 @@ static int tsi148_slave_set(struct vme_slave_resource *image, int enabled, addr |= TSI148_LCSR_ITAT_AS_A64; break; default: dev_err(tsi148_bridge->parent, "Invalid address space\n"); return -EINVAL; - break; } /* Convert 64-bit variables to 2x 32-bit variables */ reg_split(vme_base, &vme_base_high, &vme_base_low); @@ -993,11 +992,10 @@ static int tsi148_master_set(struct vme_master_resource *image, int enabled, default: spin_unlock(&image->lock); dev_err(tsi148_bridge->parent, "Invalid address space\n"); retval = -EINVAL; goto err_aspace; - break; } temp_ctl &= ~(3<<4); if (cycle & VME_SUPER) temp_ctl |= TSI148_LCSR_OTAT_SUP; @@ -1501,11 +1499,10 @@ static int tsi148_dma_set_vme_src_attributes(struct device *dev, __be32 *attr, val |= TSI148_LCSR_DSAT_AMODE_USER4; break; default: dev_err(dev, "Invalid address space\n"); return -EINVAL; - break; } if (cycle & VME_SUPER) val |= TSI148_LCSR_DSAT_SUP; if (cycle & VME_PROG) @@ -1601,11 +1598,10 @@ static int tsi148_dma_set_vme_dest_attributes(struct device *dev, __be32 *attr, val |= TSI148_LCSR_DDAT_AMODE_USER4; break; default: dev_err(dev, "Invalid address space\n"); return -EINVAL; - break; } if (cycle & VME_SUPER) val |= TSI148_LCSR_DDAT_SUP; if (cycle & VME_PROG) @@ -1699,11 +1695,10 @@ static int tsi148_dma_list_add(struct vme_dma_list *list, break; default: dev_err(tsi148_bridge->parent, "Invalid source type\n"); retval = -EINVAL; goto err_source; - break; } /* Assume last link - this will be over-written by adding another */ entry->descriptor.dnlau = cpu_to_be32(0); entry->descriptor.dnlal = cpu_to_be32(TSI148_LCSR_DNLAL_LLA); @@ -1736,11 +1731,10 @@ static int tsi148_dma_list_add(struct vme_dma_list *list, break; default: dev_err(tsi148_bridge->parent, "Invalid destination type\n"); retval = -EINVAL; goto err_dest; - break; } /* Fill out count */ entry->descriptor.dcnt = cpu_to_be32((u32)count); @@ -1962,11 +1956,10 @@ static int tsi148_lm_set(struct vme_lm_resource *lm, unsigned long long lm_base, break; default: mutex_unlock(&lm->mtx); dev_err(tsi148_bridge->parent, "Invalid address space\n"); return -EINVAL; - break; } if (cycle & VME_SUPER) lm_ctl |= TSI148_LCSR_LMAT_SUPR ; if (cycle & VME_USER) diff --git a/drivers/vme/vme.c b/drivers/vme/vme.c index b398293980b6..e1a940e43327 100644 --- a/drivers/vme/vme.c +++ b/drivers/vme/vme.c @@ -50,27 +50,22 @@ static struct vme_bridge *find_bridge(struct vme_resource *resource) /* Get list to search */ switch (resource->type) { case VME_MASTER: return list_entry(resource->entry, struct vme_master_resource, list)->parent; - break; case VME_SLAVE: return list_entry(resource->entry, struct vme_slave_resource, list)->parent; - break; case VME_DMA: return list_entry(resource->entry, struct vme_dma_resource, list)->parent; - break; case VME_LM: return list_entry(resource->entry, struct vme_lm_resource, list)->parent; - break; default: printk(KERN_ERR "Unknown resource type\n"); return NULL; - break; } } /** * vme_free_consistent - Allocate contiguous memory. @@ -177,26 +172,22 @@ size_t vme_get_size(struct vme_resource *resource) &aspace, &cycle, &dwidth); if (retval) return 0; return size; - break; case VME_SLAVE: retval = vme_slave_get(resource, &enabled, &base, &size, &buf_base, &aspace, &cycle); if (retval) return 0; return size; - break; case VME_DMA: return 0; - break; default: printk(KERN_ERR "Unknown resource type\n"); return 0; - break; } } EXPORT_SYMBOL(vme_get_size); int vme_check_window(u32 aspace, unsigned long long vme_base, diff --git a/drivers/watchdog/geodewdt.c b/drivers/watchdog/geodewdt.c index 83418924e30a..0b699c783d57 100644 --- a/drivers/watchdog/geodewdt.c +++ b/drivers/watchdog/geodewdt.c @@ -148,12 +148,10 @@ static long geodewdt_ioctl(struct file *file, unsigned int cmd, switch (cmd) { case WDIOC_GETSUPPORT: return copy_to_user(argp, &ident, sizeof(ident)) ? -EFAULT : 0; - break; - case WDIOC_GETSTATUS: case WDIOC_GETBOOTSTATUS: return put_user(0, p); case WDIOC_SETOPTIONS: diff --git a/fs/efs/inode.c b/fs/efs/inode.c index 89e73a6f0d36..64f3a54a0f72 100644 --- a/fs/efs/inode.c +++ b/fs/efs/inode.c @@ -161,11 +161,10 @@ struct inode *efs_iget(struct super_block *super, unsigned long ino) init_special_inode(inode, inode->i_mode, device); break; default: pr_warn("unsupported inode mode %o\n", inode->i_mode); goto read_inode_error; - break; } unlock_new_inode(inode); return inode; diff --git a/fs/ocfs2/cluster/tcp.c b/fs/ocfs2/cluster/tcp.c index 79a231719460..3bd8119bed5e 100644 --- a/fs/ocfs2/cluster/tcp.c +++ b/fs/ocfs2/cluster/tcp.c @@ -1196,11 +1196,10 @@ static int o2net_process_message(struct o2net_sock_container *sc, break; default: msglog(hdr, "bad magic\n"); ret = -EINVAL; goto out; - break; } /* find a handler for it */ handler_status = 0; nmh = o2net_handler_get(be16_to_cpu(hdr->msg_type), diff --git a/kernel/bpf/syscall.c b/kernel/bpf/syscall.c index 1110ecd7d1f3..8f50c9c19f1b 100644 --- a/kernel/bpf/syscall.c +++ b/kernel/bpf/syscall.c @@ -2911,11 +2911,10 @@ attach_type_to_prog_type(enum bpf_attach_type attach_type) { switch (attach_type) { case BPF_CGROUP_INET_INGRESS: case BPF_CGROUP_INET_EGRESS: return BPF_PROG_TYPE_CGROUP_SKB; - break; case BPF_CGROUP_INET_SOCK_CREATE: case BPF_CGROUP_INET_SOCK_RELEASE: case BPF_CGROUP_INET4_POST_BIND: case BPF_CGROUP_INET6_POST_BIND: return BPF_PROG_TYPE_CGROUP_SOCK; diff --git a/security/integrity/ima/ima_appraise.c b/security/integrity/ima/ima_appraise.c index 3dd8c2e4314e..f400a6122b3c 100644 --- a/security/integrity/ima/ima_appraise.c +++ b/security/integrity/ima/ima_appraise.c @@ -179,11 +179,10 @@ enum hash_algo ima_get_hash_algo(struct evm_ima_xattr_data *xattr_value, case EVM_IMA_XATTR_DIGSIG: sig = (typeof(sig))xattr_value; if (sig->version != 2 || xattr_len <= sizeof(*sig)) return ima_hash_algo; return sig->hash_algo; - break; case IMA_XATTR_DIGEST_NG: /* first byte contains algorithm id */ ret = xattr_value->data[0]; if (ret < HASH_ALGO__LAST) return ret; diff --git a/security/keys/trusted-keys/trusted_tpm1.c b/security/keys/trusted-keys/trusted_tpm1.c index b9fe02e5f84f..eddc9477d42a 100644 --- a/security/keys/trusted-keys/trusted_tpm1.c +++ b/security/keys/trusted-keys/trusted_tpm1.c @@ -899,11 +899,10 @@ static int datablob_parse(char *datablob, struct trusted_key_payload *p, return ret; ret = Opt_update; break; case Opt_err: return -EINVAL; - break; } return ret; } static struct trusted_key_options *trusted_options_alloc(void) diff --git a/security/safesetid/lsm.c b/security/safesetid/lsm.c index 8a176b6adbe5..1079c6d54784 100644 --- a/security/safesetid/lsm.c +++ b/security/safesetid/lsm.c @@ -123,11 +123,10 @@ static int safesetid_security_capable(const struct cred *cred, * set*uid() (e.g. setting up userns uid mappings). */ pr_warn("Operation requires CAP_SETUID, which is not available to UID %u for operations besides approved set*uid transitions\n", __kuid_val(cred->uid)); return -EPERM; - break; case CAP_SETGID: /* * If no policy applies to this task, allow the use of CAP_SETGID for * other purposes. */ @@ -138,15 +137,13 @@ static int safesetid_security_capable(const struct cred *cred, * set*gid() (e.g. setting up userns gid mappings). */ pr_warn("Operation requires CAP_SETGID, which is not available to GID %u for operations besides approved set*gid transitions\n", __kuid_val(cred->uid)); return -EPERM; - break; default: /* Error, the only capabilities were checking for is CAP_SETUID/GID */ return 0; - break; } return 0; } /* diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 869af8a32c98..4eabece4dcba 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -466,11 +466,10 @@ static int snd_rme32_capture_getrate(struct rme32 * rme32, int *is_adat) return 44100; case 7: return 32000; default: return -1; - break; } else switch (n) { /* supporting the CS8412 */ case 0: return -1; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4a1f576dd9cf..3382c069fd3d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2284,11 +2284,10 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) switch (hdspm->io_type) { case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 16) & 0xF; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); return (status >> HDSPM_AES32_wcFreq_bit) & 0xF; default: break; @@ -2310,11 +2309,10 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) switch (hdspm->io_type) { case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 20) & 0xF; - break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); return (status >> 1) & 0xF; default: break; @@ -2336,11 +2334,10 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) switch (hdspm->io_type) { case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_RD_STATUS_2); return (status >> 12) & 0xF; - break; default: break; } } @@ -2356,11 +2353,10 @@ static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) switch (hdspm->io_type) { case AES32: timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); return (timecode >> (4*index)) & 0xF; - break; default: break; } return 0; } @@ -3843,22 +3839,20 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) return 2; else return 1; } return 0; - break; case MADI: status2 = hdspm_read(hdspm, HDSPM_statusRegister2); if (status2 & HDSPM_wcLock) { if (status2 & HDSPM_wcSync) return 2; else return 1; } return 0; - break; case RayDAT: case AIO: status = hdspm_read(hdspm, HDSPM_statusRegister); @@ -3866,12 +3860,10 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) return 2; else if (status & 0x1000000) return 1; return 0; - break; - case MADIface: break; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 7ab10028d9fa..012fbec5e6a7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -730,38 +730,31 @@ static inline int rme9652_spdif_sample_rate(struct snd_rme9652 *s) rate_bits = rme9652_read(s, RME9652_status_register) & RME9652_F; switch (rme9652_decode_spdif_rate(rate_bits)) { case 0x7: return 32000; - break; case 0x6: return 44100; - break; case 0x5: return 48000; - break; case 0x4: return 88200; - break; case 0x3: return 96000; - break; case 0x0: return 64000; - break; default: dev_err(s->card->dev, "%s: unknown S/PDIF input rate (bits = 0x%x)\n", s->card_name, rate_bits); return 0; - break; } } /*----------------------------------------------------------------------------- Control Interface diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c index 1be82113c59a..817d8259758c 100644 --- a/sound/soc/codecs/wcd-clsh-v2.c +++ b/sound/soc/codecs/wcd-clsh-v2.c @@ -478,11 +478,10 @@ static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); break; case WCD_CLSH_STATE_HPHR: wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); break; - break; case WCD_CLSH_STATE_LO: wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); break; default: break; diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index c56b9329240f..d8ced4559bf2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -309,11 +309,10 @@ static int wl1273_startup(struct snd_pcm_substream *substream, return -EINVAL; } break; default: return -EINVAL; - break; } return 0; } diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index bbe8d782e0af..b1ca64d2f7ea 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -500,11 +500,10 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, */ ret = skl_decoupled_trigger(substream, cmd); if (ret < 0) return ret; return skl_run_pipe(skl, mconfig->pipe); - break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: /* diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a6b72ad53b43..2d85cc4c67fb 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2383,11 +2383,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); case -EPROBE_DEFER: goto err; - break; } if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err;

12 18

ALSA error on finding "working profile" for a USB MIDI device.
by GitHub issues - edited 20 Oct '20

20 Oct '20

alsa-project/alsa-lib issue #89 was edited from radiobla: TL;DR: After upgrading alsa-* packages yesterday, on the next reboot, systemd (apparently, pulseaudio.service) begun to show an error message related to "module-alsa-card" not finding a "working profile" for a USB MIDI device I have connected. Despite the error message, the device is working fine (through "raw MIDI" and also with jack2). Anyway, I want go further and find the root cause of this problem, since it's very annoying to see the Sound Service failing on every reboot. Here is my pacman.log from yesterday: **[ALPM] upgraded alsa-topology-conf (1.2.3-1 -> 1.2.4-1) [ALPM] upgraded alsa-ucm-conf (1.2.3-2 -> 1.2.4-1) [ALPM] upgraded alsa-lib (1.2.3.2-1 -> 1.2.4-1) [ALPM] upgraded alsa-utils (1.2.3-2 -> 1.2.4-1) [ALPM] upgraded lib32-alsa-lib (1.2.3.2-1 -> 1.2.4-1)** After a reboot, the Sound Service could start, but begun to show an error message. Taking a look on the systemd logs with journalctl -xe --user-unit pulseaudio I found this: systemd[1027]: Starting Sound Service... ... **pulseaudio[1037]: Failed to find a working profile. pulseaudio[1037]: Failed to load module "module-alsa-card" (argument: "device_id="3" name="usb-Roland_A-> pulseaudio[1037]: module-rescue-stream is obsolete and should no longer be loaded. Please remove it from> systemd[1027]: Started Sound Service.** I disabled the "load-module module-rescue-streams" on the default.pa, but the other issue persists. They seem not be related, IMO. I wanted to rule out that chance anyway. By stopping pulseaudio service through systemd and running pulseaudio -vvv manually, I got more verbose information. **E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Roland_A-Series_Keyboard-00" card_name="alsa_card.usb-Roland_A-Series_Keyboard-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:14.0/usb1/1-10/1-10:1.0/sound/card1 (alsa_card.usb-Roland_A-Series_Keyboard-00) failed to load module.** Regarding specifically to the following error message... **E: [pulseaudio] module-alsa-card.c: Failed to find a working profile.** What does it mean it is not finding a "working profile"? How could I investigate it further? It worth mentioning that the device (an USB Keyboard Controller) is working perfectly with jack2. I tested it with an "aseqdump". $ aseqdump -l Port Client name Port name 0:0 System Timer 0:1 System Announce 14:0 Midi Through Midi Through Port-0 20:0 A-Series Keyboard A-Series Keyboard Keyboard $ aseqdump -p20:0 Waiting for data. Press Ctrl+C to end. Source Event Ch Data 20:0 Note on 0, note 41, velocity 76 20:0 Note on 0, note 40, velocity 82 Although the device is working even with the error message, it's very annoying to see a "failed service" on initialization. Thanks! More occurrences of this error were provided on Arch Linux [forum](https://bbs.archlinux.org/viewtopic.php?pid=1932801) Issue URL : https://github.com/alsa-project/alsa-lib/issues/89 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA error on finding "working profile" for a USB MIDI device.
by GitHub issues - opened 20 Oct '20

20 Oct '20

alsa-project/alsa-lib issue #89 was opened from radiobla: TL;DR: After upgrading alsa-* packages yesterday, on the next reboot, systemd (apparently, pulseaudio.service) begun to show an error message related to "module-alsa-card" not finding a "working profile" for a USB MIDI device I have connected. Despite the error message, the device is working fine (through "raw MIDI" and also with jack2). Anyway, I want go further and find the root cause of this problem, since it's very annoying to see the Sound Service failing on every reboot. Here is my pacman.log from yesterday: **[ALPM] upgraded alsa-topology-conf (1.2.3-1 -> 1.2.4-1) [ALPM] upgraded alsa-ucm-conf (1.2.3-2 -> 1.2.4-1) [ALPM] upgraded alsa-lib (1.2.3.2-1 -> 1.2.4-1) [ALPM] upgraded alsa-utils (1.2.3-2 -> 1.2.4-1) [ALPM] upgraded lib32-alsa-lib (1.2.3.2-1 -> 1.2.4-1)** After a reboot, the Sound Service could start, but begun to show an error message. Taking a look on the systemd logs with journalctl -xe --user-unit pulseaudio I found this: systemd[1027]: Starting Sound Service... ... **pulseaudio[1037]: Failed to find a working profile. pulseaudio[1037]: Failed to load module "module-alsa-card" (argument: "device_id="3" name="usb-Roland_A-> pulseaudio[1037]: module-rescue-stream is obsolete and should no longer be loaded. Please remove it from> systemd[1027]: Started Sound Service.** I disabled the "load-module module-rescue-streams" on the default.pa, but the other issue persists. They seem not be related, IMO. I wanted to rule out that chance anyway. By stopping pulseaudio service through systemd and running pulseaudio -vvv manually, I got more verbose information. **E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-Roland_A-Series_Keyboard-00" card_name="alsa_card.usb-Roland_A-Series_Keyboard-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:14.0/usb1/1-10/1-10:1.0/sound/card1 (alsa_card.usb-Roland_A-Series_Keyboard-00) failed to load module.** Regarding specifically to the following error message... **E: [pulseaudio] module-alsa-card.c: Failed to find a working profile.** What does it mean it is not finding a "working profile"? How could I investigate it further? It worth mentioning that the device (an USB Keyboard Controller) is working perfectly with jack2. I tested it with an "aseqdump". $ aseqdump -l Port Client name Port name 0:0 System Timer 0:1 System Announce 14:0 Midi Through Midi Through Port-0 20:0 A-Series Keyboard A-Series Keyboard Keyboard $ aseqdump -p20:0 Waiting for data. Press Ctrl+C to end. Source Event Ch Data 20:0 Note on 0, note 41, velocity 76 20:0 Note on 0, note 40, velocity 82 Although the device is working even with the error message, it's very annoying to see a "failed service" on initialization. Thanks! More occurrences of this error were provided on Arch Linux forum: [](https://bbs.archlinux.org/viewtopic.php?pid=1932801) Issue URL : https://github.com/alsa-project/alsa-lib/issues/89 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] Line6 Pod Go interface requires static clock rate quirk
by Lukasz Halman 20 Oct '20

20 Oct '20

Recently released Line6 Pod Go requires static clock rate quirk to make its usb audio interface working. Added its usb id to the list of similar line6 devices. Signed-off-by: Lukasz Halman <lukasz.halman(a)gmail.com> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 1b28d01d1..3bfead393 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -406,6 +406,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0e41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0e41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0e41, 0x4247): /* Line6 Pod Go */ case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ -- 2.25.1

2 1

[PATCH] ASoC: cs42l51: manage mclk shutdown delay
by Olivier Moysan 20 Oct '20

20 Oct '20

A delay must be introduced before the shutdown down of the mclk, as stated in CS42L51 datasheet. Otherwise the codec may produce some noise after the end of DAPM power down sequence. The delay between DAC and CLOCK_SUPPLY widgets is too short. Add a delay in mclk shutdown request to manage the shutdown delay explicitly. From experiments, at least 10ms delay is necessary. Set delay to 20ms as recommended in Documentation/timers/timers-howto.rst when using msleep(). Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Recommended Power-Down Sequence: (see https://statics.cirrus.com/pubs/proDatasheet/CS42L51_F2.pdf) 1. Mute the DACs and ADCs. 2. Disable soft ramp and zero-cross volume transitions 3. Set the PDN bit to 1. 4. Wait at least 100 μs. The codec will be fully powered down after this 100 μs delay. Prior to the removal of the master clock(MCLK), this delay of at least 100 μs must be implemented after step 3 to avoid premature disruption of the codec’s power down sequence. --- sound/soc/codecs/cs42l51.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 097c4e8d9950..d151a1aa313e 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -254,8 +254,25 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + if (SND_SOC_DAPM_EVENT_ON(event)) + return clk_prepare_enable(cs42l51->mclk_handle); + + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { -- 2.17.1

2 1

[PATCH] ASoC: cs35l3x: Use neutral language in amp drivers
by David Rhodes 20 Oct '20

20 Oct '20

Revise variable names and comments in cs35l35 and cs35l36 amp drivers. Signed-off-by: David Rhodes <david.rhodes(a)cirrus.com> --- sound/soc/codecs/cs35l35.c | 10 +++++----- sound/soc/codecs/cs35l35.h | 2 +- sound/soc/codecs/cs35l36.c | 10 +++++----- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index e330427..3774b69 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -372,12 +372,12 @@ static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_CBM_CFM: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = false; + cs35l35->clock_mode = false; break; case SND_SOC_DAIFMT_CBS_CFS: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = true; + cs35l35->clock_mode = true; break; default: return -EINVAL; @@ -556,8 +556,8 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, } sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; - /* Only certain ratios are supported in I2S Slave Mode */ - if (cs35l35->slave_mode) { + /* Only certain ratios are supported in I2S Device Mode */ + if (cs35l35->clock_mode) { switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_48FS: @@ -568,7 +568,7 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { - /* Only certain ratios supported in I2S MASTER Mode */ + /* Only certain ratios supported in I2S Controller Mode */ switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_64FS: diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index ffb154c..017fa6d 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -282,7 +282,7 @@ struct cs35l35_private { int sclk; bool pdm_mode; bool i2s_mode; - bool slave_mode; + bool clock_mode; /* GPIO for /RST */ struct gpio_desc *reset_gpio; struct completion pdn_done; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76..86f5533 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -756,14 +756,14 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, { struct cs35l36_private *cs35l36 = snd_soc_component_get_drvdata(component_dai->component); - unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, clock_mode, clk_frc; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - slave_mode = 1; + clock_mode = 1; break; case SND_SOC_DAIFMT_CBS_CFS: - slave_mode = 0; + clock_mode = 0; break; default: return -EINVAL; @@ -771,10 +771,10 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, CS35L36_SCLK_MSTR_MASK, - slave_mode << CS35L36_SCLK_MSTR_SHIFT); + clock_mode << CS35L36_SCLK_MSTR_SHIFT); regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, CS35L36_LRCLK_MSTR_MASK, - slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + clock_mode << CS35L36_LRCLK_MSTR_SHIFT); switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { case SND_SOC_DAIFMT_CONT: -- 1.9.1

2 1

ASoC: error at snd_soc_pcm_dai_trigger on acp_rn_pdm_dma.0
by GitHub issues - opened 20 Oct '20

20 Oct '20

alsa-project/alsa-lib issue #88 was opened from JcdeA: I am on Artix and this has been an issue for a while. Pauvcontrol shows Dummy Output as the only output available. https://www.reddit.com/r/archlinux/comments/e73u68/dummy_audio_output_after… I saw this post and downgraded alsa-lib-1.2.4 to alsa-lib-1.2.3 and it works perfectly now. I tried all of the solutions on the fourms, but none worked except downgrading. Seems like a issue with the latest version. Issue URL : https://github.com/alsa-project/alsa-lib/issues/88 Repository URL: https://github.com/alsa-project/alsa-lib

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Re: [PATCH v2 2/2] ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation
by Rob Herring 20 Oct '20

20 Oct '20

On Wed, Sep 23, 2020 at 9:40 PM Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: > > > Hi Rob > > > > From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > > > > > This patch switches from .txt base to .yaml base Document. > > > It is still keeping detail explanations at .txt > > > > > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > > --- > (snip) > > > + # For OF-graph > > > + port: > > > + description: OF-Graph subnode > > > + $ref: "audio-graph-card.yaml#/definitions/port" > > > > Something like this should work (at the top-level): > > > > allOf: > > - $ref: "audio-graph-card.yaml# > > Hmm.. it will be error, see below (A). > How to solve it ?? All the 'allOf's have to be grouped. > (I will keep to use above style) > > > > +# use patternProperties to avoid naming "xxx,yyy" > > > +patternProperties: > > > + "^rcar_sound,dvc$": > > > > None of these are patterns. > > As always happen, "xxx,yyy" style will be error for me without patternProperties. > What is wrong ?? Node names shouldn't really have vendor prefixes, but since you already depend on this we shouldn't change that. We'll need to relax the meta-schema, but for now it's fine as-is. Rob

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[PATCH 0/7] Add dts for Rpi4 + Cirrus Lochnagar and codecs
by Richard Fitzgerald 19 Oct '20

19 Oct '20

This set of patches provides support for using the Cirrus Logic Lochnagar audio development platform plus Cirrus Logic Madera/Arizona codecs with the simple-card machine driver and a Raspberry Pi4. The ultimate aim is to provide the dts file but some updates are needed to the simple-card machine driver. Patches are needed to simple-card to enable support for setting the component sysclks and plls. The codec sysclks and plls cannot be placed under the clock framwork because they are I2C/SPI-connected peripherals and access to the registers would cause a nested get of the I2C/SPI bus clock. The clock framework does not support this and it would result in a deadlock. Richard Fitzgerald (7): of: base: Add of_count_phandle_with_fixed_args() ASoC: simple-card: Add plls and sysclks DT schema ASoC: simple-card: Support setting component plls and sysclks ASoC: arizona: Allow codecs to be selected from kernel config ASoC: madera: Allow codecs to be selected from kernel config ARM: dts: Add dts for Raspberry Pi 4 + Cirrus Logic Lochnagar2 MAINTAINERS: Add dts for Cirrus Logic Lochnagar on RPi4 .../bindings/sound/simple-card.yaml | 56 + MAINTAINERS | 1 + arch/arm/boot/dts/Makefile | 1 + .../dts/bcm2711-rpi4b-cirrus-lochnagar.dts | 1296 +++++++++++++++++ drivers/of/base.c | 42 + include/linux/of.h | 9 + include/sound/simple_card_utils.h | 24 + sound/soc/codecs/Kconfig | 18 +- sound/soc/generic/simple-card-utils.c | 184 +++ sound/soc/generic/simple-card.c | 14 +- 10 files changed, 1635 insertions(+), 10 deletions(-) create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar.dts -- 2.20.1

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[PATCH 1/1] ASoC: dt-bindings: stm32: convert audio dfsdm to json-schema
by Olivier Moysan 19 Oct '20

19 Oct '20

Convert the STM32 DFSDM audio bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- .../bindings/sound/st,stm32-adfsdm.txt | 63 ------------------- .../bindings/sound/st,stm32-adfsdm.yaml | 42 +++++++++++++ 2 files changed, 42 insertions(+), 63 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt deleted file mode 100644 index 864f5b00b031..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt +++ /dev/null @@ -1,63 +0,0 @@ -STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM) - -The DFSDM allows PDM microphones capture through SPI interface. The Audio -interface is seems as a sub block of the DFSDM device. -For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt - -Required properties: - - compatible: "st,stm32h7-dfsdm-dai". - - - #sound-dai-cells : Must be equal to 0 - - - io-channels : phandle to iio dfsdm instance node. - -Example of a sound card using audio DFSDM node. - - sound_card { - compatible = "audio-graph-card"; - - dais = <&cpu_port>; - }; - - dfsdm: dfsdm@40017000 { - compatible = "st,stm32h7-dfsdm"; - reg = <0x40017000 0x400>; - clocks = <&rcc DFSDM1_CK>; - clock-names = "dfsdm"; - #interrupt-cells = <1>; - #address-cells = <1>; - #size-cells = <0>; - - dfsdm_adc0: filter@0 { - compatible = "st,stm32-dfsdm-dmic"; - reg = <0>; - interrupts = <110>; - dmas = <&dmamux1 101 0x400 0x00>; - dma-names = "rx"; - st,adc-channels = <1>; - st,adc-channel-names = "dmic0"; - st,adc-channel-types = "SPI_R"; - st,adc-channel-clk-src = "CLKOUT"; - st,filter-order = <5>; - - dfsdm_dai0: dfsdm-dai { - compatible = "st,stm32h7-dfsdm-dai"; - #sound-dai-cells = <0>; - io-channels = <&dfsdm_adc0 0>; - cpu_port: port { - dfsdm_endpoint: endpoint { - remote-endpoint = <&dmic0_endpoint>; - }; - }; - }; - }; - - dmic0: dmic@0 { - compatible = "dmic-codec"; - #sound-dai-cells = <0>; - port { - dmic0_endpoint: endpoint { - remote-endpoint = <&dfsdm_endpoint>; - }; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.yaml b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.yaml new file mode 100644 index 000000000000..d953ec524ba2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-adfsdm.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-adfsdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The DFSDM allows PDM microphones capture through the SPI interface. + The Audio interface is seen as a sub block of the DFSDM device. + For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.yaml + +properties: + compatible: + enum: + - st,stm32h7-dfsdm-dai + + "#sound-dai-cells": + const: 0 + + io-channels: + description: phandle to iio dfsdm instance node + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - io-channels + +examples: + - | + asoc_pdm0: dfsdm-dai { + compatible = "st,stm32h7-dfsdm-dai"; + #sound-dai-cells = <0>; + io-channels = <&dfsdm0 0>; + }; + +... -- 2.17.1

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