October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 04 Oct '19

04 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Based on for-next branch from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound Changes since v2: - none. Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Don't generate kcontrols without READ flags
by Charles Keepax 04 Oct '19

04 Oct '19

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> User space always expects to be able to read ALSA controls, so ensure no kcontrols are generated without an appropriate READ flag. In the case of a read of such a control zeros will be returned. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 85396d920e0a9..9b8bb7bbe945d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoc: rockchip: i2s: Fix RPM imbalance
by Robin Murphy 04 Oct '19

04 Oct '19

If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait for an absent DMA channel, we return without disabling RPM, which makes subsequent re-probe attempts scream with errors about the unbalanced enable. Don't do that. Fixes: ebb75c0bdba2 ("ASoC: rockchip: i2s: Adjust devm usage") Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af2d5a6124c8..61c984f10d8e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; -- 2.21.0.dirty

2 1

[alsa-devel] [PATCH 0/6][resend] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 04 Oct '19

04 Oct '19

Hi Mark Some soc-core cleanup patch step3 are accepted, but not all. These are remains. Kuninori Morimoto (6): ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() sound/soc/soc-core.c | 109 +++++++++++++++++++++++++++------------------------ sound/soc/soc-ops.c | 11 +----- 2 files changed, 59 insertions(+), 61 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

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[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 04 Oct '19

04 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH] ASoC: rsnd: Document r8a774b1 bindings
by Biju Das 04 Oct '19

04 Oct '19

Document SoC specific bindings for RZ/G2N (r8a774b1) SoC. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5c52182..797fd03 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -268,6 +268,7 @@ Required properties: - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a77470" (RZ/G1C) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) + - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) -- 2.7.4

2 1

Re: [alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by Kai Vehmanen 04 Oct '19

04 Oct '19

Hi, (routing back to alsa-devel, Ed's alsa-devel subscription is still pending so he can't sent to the list, forwarding with his permission) On Fri, 4 Oct 2019, ed nwave wrote: > The example used of a voice changing app was just an illustration of how > I need to connect "my driver" directly to a sound card driver that's > all. I understand the user-space comment entirely. The actual > objective is one common driver ( my PCM driver ) being used by 4 music > applications where I can dynamically switch the output PCM without > restarting the applications or the applications even knowing the output > cards not real. so far this is pretty much directly from original design goals of JACK. But ok, then you mention this: > I must have direct access to audio hardware i.e. hw:1,0 > for applications like RoonReady/ RAAT. So Pulseaudio's out. I require > the ability to Now this can be a problem of course if you cannot modify the apps. The you would have to route this via alsa-lib plugins. > follow ( like and MPD playlist ). As I understand Jack is set for a > certain rate and format at the start when connecting and does not update > rates automatically if the rate of the songs change; can jack connect > directly to hw and handle DSD rates?. This is all part of a headless I've been out of the loop for some years now, but no, I don't think JACK supports these. But you are anyways customizing quite a lot of SW, so you could e.g. consider extending JACK and/or some other audio server for this purpose, or just write a new one. > Other requirements are my applications must have no plugins or dmix and > connect directly to hardware i.e. hw:1,0 - due to rate demands, no > remixing. Another requirement is the latency required by one company to > attain multi-room streaming being about 1ms max, this is for > certification. Finally there's the need to change bit rate settings I think reaching the latency requirement (and not doing mixing) can certainly be done in user-space, so I still read the "need to access harwdare directly" more as "existing user-space components are not suitable for my needs" than a strict requirement for doing this in kernel. But alas, I get your context and I understand these constraints are real for you as well. Just wanted to make sure you have taken good enough look at user-space options. > All that's left for me to do is to connect pipe the data directly to the > driver for the my sound card. Question is can it be done? And if so, > where do I even start? How can I import snd_usb_audio and open it so it > appears a the typical snd_card struct? The longrunning trend has been to push stuff like this to user-space, so you will not find much examples. I haven't done this myself, so I'll let other people make recommendations. I'd start by looking at ALSA OSS emulation and the virtualized drivers we have (like sound/xen/). Br, Kai

1 0

Re: [alsa-devel] kernel freeze regression relating to imx-sdma
by Kevin Groeneveld 04 Oct '19

04 Oct '19

Hi Lucas, Thanks for the feedback. >-----Original Message----- >From: Lucas Stach [mailto:l.stach@pengutronix.de] >Sent: October 3, 2019 5:26 PM >To: Kevin Groeneveld; dmaengine(a)vger.kernel.org >Subject: Re: kernel freeze regression relating to imx-sdma > >Hi Kevin, > >Am Donnerstag, den 03.10.2019, 21:11 +0000 schrieb Kevin Groeneveld: >> We have recently come across an issue which can cause the linux kernel to >> freeze on an iMX6 based system. After bisecting the issue I have found the >> issue is related to the following imx-sdma commits: >> >> 64068853bc77786d1a28abb4087d6a3e93aedbe2 dmaengine: imx-sdma: use GFP_NOWAIT for dma descriptor allocations >> b8603d2a5795c42f78998e70dc792336e0dc20c9 dmaengine: imx-sdma: implement channel termination via worker >> ebb853b1bd5f659b92c71dc6a9de44cfc37c78c0 Revert "dmaengine: imx-sdma: alloclate bd memory from dma pool" >> c06abca69218ac42fa58d1ba7a7b0d9bab5f1b18 Revert "dmaengine: imx-sdma: Use GFP_NOWAIT for dma allocations" >> >> If I revert these four commits the freeze does not happen. The issue is still >> present up to at least v5.3. >> >> The problem seems to be related to overruns when doing an SPDIF capture, >> particularly when there is not actually a valid SPDIF input to the iMX6. When >> the issue occurs the following is (usually) printed on the console (using a v5.0 >> based kernel): >> >> [ 1251.944877] rcu: INFO: rcu_sched detected stalls on CPUs/tasks: >> [ 1251.950823] rcu: (detected by 0, t=2102 jiffies, g=220425, q=0) >> [ 1251.956866] rcu: All QSes seen, last rcu_sched kthread activity 2102 (95117-93015), jiffies_till_next_fqs=1, root ->qsmask 0x0 >> [ 1251.968266] arecord S 0 1916 1908 0x00000002 >> [ 1251.973816] [<8010dcec>] (unwind_backtrace) from [<8010a4c4>] (show_stack+0x10/0x14) >> [ 1251.981596] [<8010a4c4>] (show_stack) from [<8016b314>] (rcu_check_callbacks+0x57c/0x5d8) >> [ 1251.989806] [<8016b314>] (rcu_check_callbacks) from [<8016fe84>] (update_process_times+0x30/0x5c) >> [ 1251.998709] [<8016fe84>] (update_process_times) from [<8017e960>] (tick_sched_timer+0x44/0x94) >> [ 1252.007353] [<8017e960>] (tick_sched_timer) from [<80170a84>] (__hrtimer_run_queues+0xd8/0x160) >> [ 1252.016081] [<80170a84>] (__hrtimer_run_queues) from [<8017103c>] (hrtimer_interrupt+0xdc/0x284) >> [ 1252.024891] [<8017103c>] (hrtimer_interrupt) from [<8010d6c0>] (twd_handler+0x2c/0x38) >> [ 1252.032839] [<8010d6c0>] (twd_handler) from [<80160b34>] (handle_percpu_devid_irq+0x70/0x11c) >> [ 1252.041401] [<80160b34>] (handle_percpu_devid_irq) from [<8015c234>] (generic_handle_irq+0x18/0x28) >> [ 1252.050480] [<8015c234>] (generic_handle_irq) from [<8015c814>] (__handle_domain_irq+0xa0/0xb4) >> [ 1252.059211] [<8015c814>] (__handle_domain_irq) from [<8039098c>] (gic_handle_irq+0x58/0x90) >> [ 1252.067590] [<8039098c>] (gic_handle_irq) from [<80101a0c>] (__irq_svc+0x6c/0x90) >> [ 1252.075085] Exception stack(0x848afe30 to 0x848afe78) >> [ 1252.080155] fe20: 8decf200 00000000 00000000 80b84ac8 >> [ 1252.088354] fe40: 00000000 8decf200 892f7800 00001589 892f7800 00000000 00000000 848ae000 >> [ 1252.096549] fe60: 8d8cf000 848afe80 805721e8 8056aae4 60000013 ffffffff >> [ 1252.103200] [<80101a0c>] (__irq_svc) from [<8056aae4>] (__snd_pcm_stream_unlock_mode+0xc0/0xd8) >> [ 1252.111925] [<8056aae4>] (__snd_pcm_stream_unlock_mode) from [<00000000>] ( (null)) >> [ 1252.119695] rcu: rcu_sched kthread starved for 2102 jiffies! g220425 f0x2 RCU_GP_WAIT_FQS(5) ->state=0x0 ->cpu=0 >> [ 1252.129877] rcu: RCU grace-period kthread stack dump: >> [ 1252.134938] rcu_sched R running task 0 10 2 0x00000000 >> [ 1252.142034] [<806eee5c>] (__schedule) from [<806eef80>] (schedule+0x90/0xa0) >> [ 1252.149116] [<806eef80>] (schedule) from [<806f2244>] (schedule_timeout+0x1f0/0x238) >> [ 1252.156891] [<806f2244>] (schedule_timeout) from [<8016940c>] (rcu_gp_kthread+0x550/0x8f0) >> [ 1252.165183] [<8016940c>] (rcu_gp_kthread) from [<8013a358>] (kthread+0x110/0x128) >> [ 1252.172693] [<8013a358>] (kthread) from [<801010e8>] (ret_from_fork+0x14/0x2c) >> [ 1252.179925] Exception stack(0x8f877fb0 to 0x8f877ff8) >> [ 1252.184991] 7fa0: 00000000 00000000 00000000 00000000 >> [ 1252.193187] 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 >> [ 1252.201380] 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 >> >> I suspect the issue may not actually be a bug with imx-sdma but a >> race condition elsewhere that is exposed by the four mentioned >> commits. >> >> I did notice that when an SPDIF overrun occurs >> dmaengine_terminate_async is called but dmaengine_synchronize is >> not. Should these calls always be paired? Or is it okay to call >> dmaengine_terminate_async without dmaengine_synchronize in some >> cases? > >Yes, it seems this uncovered a bug in the SPDIF driver. Before those >commits the dmaengine_terminate_async in the SDMA driver wasn't really >async, now it is. The API requires that you call dmaengine_synchronize >eventually after the terminate_async before starting any new operations >on the dmaengine. It doesn't necessarily need to be in the same >codepath, but the the async terminate always needs to be paired up with >a synchronize. I don't think the bug is limited to SPDIF. The calls to dmaengine_terminate_async are in sound/core/pcm_dmaengine.c. When the ALSA device is closed it does call dmaengine_synchronize. But when there is an overrun it does not. It was commit bc0e7345168c0f7483d2d1da86285d89136417cd (by Lars-Peter Clausen) where the call to dmaengine_synchronize was added for closing an ALSA device. It seems the overrun case was overlooked? I think maybe I should copy the ALSA list on this as well. I have also found another manifestation of the imx-sdma change which may be useful for others trying to debug this. If I run the following command: arecord -Dhw:4,0 -fS24_LE -c2 -r44100 --buffer-size 100 > /dev/null With kernel v4.20-rc5 I get an endless stream of overrun messages. With kernel v4.20-rc6 (which includes the four imx-sdma commits mentioned above) I get a few overrun messages but then arecord exits with "arecord: pcm_read:2096: read error: Input/output error" Best regards, Kevin

1 0

[alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by ed nwave 04 Oct '19

04 Oct '19

Hello, A little bit of an unorthodox request but I need my custom PCM driver to connect directly to another PCM driver if possible. I have an application that already sees my custom PCM driver as the connected sound card, but I also need to pipe the PCM data within my driver after altering it out to another PCM of my choice ( my actual sound card ). As an example, my PCM driver acts as a sound card connected to a microphone capture application, within the my custom PCM I've added various "dalek" and other voice changing alterations and now I need to output this stream into my actual sound card. The sound card I need my driver to import or connect to is a DAC over USB and I believe the loaded kernel module is snd_sub_audio which I can see via lsmod. It's PCM hw:1,0. So is this possible? An if so, how? Even the most basic information would be appreciated. Thanks.

2 1

[alsa-devel] Making alsamixer configurable, adding mouse support + minor fixes ...
by GitHub pull_request - opened 03 Oct '19

03 Oct '19

alsa-project/alsa-utils pull request #28 was opened from braph: ... continued from PR #26 ... Request URL : https://github.com/alsa-project/alsa-utils/pull/28 Patch URL : https://github.com/alsa-project/alsa-utils/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH v2] FROMLIST: drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 03 Oct '19

03 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

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Re: [alsa-devel] [External] Re: alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Pierre-Louis Bossart 03 Oct '19

03 Oct '19

On 9/19/19 9:54 AM, Mark Pearson wrote: >> >> Indeed UCM is required for all cases where SOF and PulseAudio are used. >> >> Our thinking was however to add this UCM file to the new repository outside >> of alsa-lib [1]. There is an on-going thread started by Jaroslav to move those >> files and relicense them as BSD-3-Clause [2] >> >> [1] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> July/153257.html >> [2] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> September/155246.html > > Thanks Pierre. > > Do we have any approximate timelines on when and how this will happen? (I'm new to this) > > One of my main aims is that we have a customer using Debian and one of our platforms that require this change - I need to make sure I understand how this would roll out and what actions they need to take in the meantime if it's not going to be available in Debian. I think the first order would be to have the file cleaned-up, with its Intel origin clearly stated with a signed-off-by tag. Then once this is done, the Debian package creation needs to be handled (using either the ALSA repo or the cloned version on SOF GitHub). I don't have any experience with Debian packages so can't really comment on the effort it would take.

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[alsa-devel] [PATCH v2 00/10] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") The main changes since v1: - Fixed bug in "ASoC: wm8994: Add support for MCLKn clock gating" patch (thanks to Charles), - dropped the MFD part patch and moved getting of clocks to the CODEC driver, - Added missing Kconfig entries to patch "ASoC: samsung: arndale: Add support for WM1811 CODEC", - Added a patch renaming the driver and a patch for exynos_defconfig. Sylwester Nawrocki (10): ASoC: wm8994: Do not register inapplicable controls for WM1811 ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Add support for setting MCLK clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: Rename Arndale card driver ARM: dts: arndale: Add audio support (WM1811 CODEC boards) ARM: exynos_defconfig: Enable Arndale audio driver .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- arch/arm/configs/exynos_defconfig | 1 + sound/soc/codecs/wm8994.c | 199 +++++++++++++--- sound/soc/codecs/wm8994.h | 10 +- sound/soc/samsung/Kconfig | 12 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/arndale.c | 217 ++++++++++++++++++ sound/soc/samsung/arndale_rt5631.c | 138 ----------- 9 files changed, 438 insertions(+), 175 deletions(-) create mode 100644 sound/soc/samsung/arndale.c delete mode 100644 sound/soc/samsung/arndale_rt5631.c -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 02 Oct '19

02 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 35 ++++++++++++ sound/hda/Kconfig | 10 +++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 97 ++++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 49 +++++----------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++----- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 173 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..700f86282a83 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_NODSP, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..1abbf74ba643 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + 0x5a98, /* Broxton-P (Appololake) */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_NODSP; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..c76c2deea47c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Do not return from probe when another DSP driver claims device"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,17 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_NODSP && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ 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_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

3 4

[alsa-devel] [PATCH 1/2] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Maciej Falkowski 02 Oct '19

02 Oct '19

Convert Samsung I2S controller to newer dt-schema format Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ------------- .../bindings/sound/samsung-i2s.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..59dc76035cb4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sangbeom Kim <sbkim73(a)samsung.com> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - "samsung,s3c6410-i2s" + - "samsung,s5pv210-i2s" + - "samsung,exynos5420-i2s" + - "samsung,exynos7-i2s" + - "samsung,exynos7-i2s1" + + reg: + maxItems: 1 + + dmas: + description: list of DMA controller phandle and DMA request line ordered pairs. + + dma-names: + description: | + identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: + - const: iis + - const: i2s_opclk0 + - items: + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + description: | + "iis" is the i2s bus clock. + For i2s1 and i2s2 - "iis", "i2s_opclk0" + For i2s0 - "iis", "i2s_opclk0", "i2s_opclk1" + + "#clock-cells": + const: 1 + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10 + &pdma0 9 + &pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>; // EXYNOS_SCLK_I2S + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

5 19

[alsa-devel] Period size setting is ignored when dmix is used
by Andrea Narciso - ART S.p.A. 02 Oct '19

02 Oct '19

I would like to set different period size (or time) values for different ALSA applications, in order to optimize latency and processor load based on specific application needs. For example, a telephony application will use a shorter period size/time than a media player application, so that low latency does not affect the quality of the call. I found out that if dmix plugin is used in the ALSA path, period size/time settings are completely ignored and forced to default values (supposedly read from the ALSA driver of the HW card). For example, not using dmix, period size value is correctly accepted (but of course I cannot play audio from multiple applications at the same time): aplay -vv --period-size=4096 -D plug:hw <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 4096 ... Adding dmix instead, I can play audio from multiple applications at the same time, but period size is ignored and I cannot therefore set different latencies for different applications: aplay -vv --period-size=4096 -D plug:dmix <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 1024 ... Am I missing something? Is there a way to implement mixing + different latency settings in ALSA applications, without using an audio server? Thanks Andrea

2 1

[alsa-devel] [PATCH] ASoC: atmel: fix atmel_ssc_set_audio link failure
by Arnd Bergmann 02 Oct '19

02 Oct '19

The ssc audio driver can call into both pdc and dma backends. With the latest rework, the logic to do this in a safe way avoiding link errors was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register' Fix it this time using Makefile hacks and a comment to prevent this from accidentally getting removed again rather than Kconfig hacks. Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 10 ++++++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed82..25c31bf64936 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,11 +10,11 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - tristate + bool depends on HAS_DMA config SND_ATMEL_SOC_DMA - tristate + bool select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 1f6890ed3738..c7d2989791be 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -6,8 +6,14 @@ snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o -obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o +# pdc and dma need to both be built-in if any user of +# ssc is built-in. +ifdef CONFIG_SND_ATMEL_SOC_PDC +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o +endif +ifdef CONFIG_SND_ATMEL_SOC_DMA +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o +endif obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o -- 2.20.0

4 6

[alsa-devel] [PATCH v3 RESEND 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

4 8

[alsa-devel] [PATCH v2 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Changes in v2: * Use C++ coment style on the entire comment in the .c files; * Do not use ternery operator; * Do not use `devm_regulator_get_optional` as the device always requires power; * Let DAPM suspend/resume the device. MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 586 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 770 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index 296de2b51c83..8ba47cac8e83 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1002,6 +1002,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ae213c6392ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -396,6 +398,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73b2d5982dcb..2118407a1dd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -305,6 +308,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..45a5d2dcc0f2 --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw +// driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..a8211362fe82 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..bf5a5d75f81a --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,586 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + st->iovdd = devm_regulator_get(st->dev, "IOVDD"); + if (IS_ERR(st->iovdd)) { + dev_err(st->dev, "Could not get iovdd: %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, + st); +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] [PATCH 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 621 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index a50e97a63bc8..77c920559d82 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -985,6 +985,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299..006b0686776c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -392,6 +394,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf4325..974dc1eda797 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -301,6 +304,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..499dd48ea01b --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw + * driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..1fd0b0287699 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..d6b90e31042f --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,621 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + if (!st->iovdd) + return 0; + + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (!st->iovdd) + return 0; + + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +#ifdef CONFIG_PM +static int adau7118_suspend(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); +} + +static int adau7118_resume(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, + SND_SOC_BIAS_STANDBY); +} +#else +#define adau7118_suspend NULL +#define adau7118_resume NULL +#endif + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .suspend = adau7118_suspend, + .resume = adau7118_resume, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + int ret = 0; + + st->iovdd = devm_regulator_get_optional(st->dev, "IOVDD"); + if (!IS_ERR(st->iovdd)) { + /* + * Only get dvdd if iovdd is present. Furthermore, + * if iovdd is present, dvdd must also be given. The two + * supplies should be controlled together since dvdd should + * not be applied before iovdd. + */ + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + ret = devm_add_action_or_reset(st->dev, + adau7118_regulator_disable, st); + } else { + if (PTR_ERR(st->iovdd) == -ENODEV) { + st->iovdd = NULL; + } else { + dev_err(st->dev, "Failed to get IOVDD %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + } + + return ret; +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 02 Oct '19

02 Oct '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

2 3

[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 02 Oct '19

02 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

1 0

[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 1a476abc723e644248dba975b71122fcf878703b make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:218:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:225:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2770.c:361:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2770.c:361:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2770.c:361:58: sparse: got restricted snd_pcm_format_t vim +211 sound/soc/codecs/tas2770.c 204 205 static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) 206 { 207 int ret; 208 struct snd_soc_component *component = tas2770->component; 209 210 switch (bitwidth) { > 211 case SNDRV_PCM_FORMAT_S16_LE: 212 ret = snd_soc_component_update_bits(component, 213 TAS2770_TDM_CFG_REG2, 214 TAS2770_TDM_CFG_REG2_RXW_MASK, 215 TAS2770_TDM_CFG_REG2_RXW_16BITS); 216 tas2770->v_sense_slot = tas2770->i_sense_slot + 2; 217 break; 218 case SNDRV_PCM_FORMAT_S24_LE: 219 ret = snd_soc_component_update_bits(component, 220 TAS2770_TDM_CFG_REG2, 221 TAS2770_TDM_CFG_REG2_RXW_MASK, 222 TAS2770_TDM_CFG_REG2_RXW_24BITS); 223 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 224 break; 225 case SNDRV_PCM_FORMAT_S32_LE: 226 ret = snd_soc_component_update_bits(component, 227 TAS2770_TDM_CFG_REG2, 228 TAS2770_TDM_CFG_REG2_RXW_MASK, 229 TAS2770_TDM_CFG_REG2_RXW_32BITS); 230 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 231 break; 232 233 default: 234 return -EINVAL; 235 } 236 237 tas2770->channel_size = bitwidth; 238 239 ret = snd_soc_component_update_bits(component, 240 TAS2770_TDM_CFG_REG5, 241 TAS2770_TDM_CFG_REG5_VSNS_MASK | 242 TAS2770_TDM_CFG_REG5_50_MASK, 243 TAS2770_TDM_CFG_REG5_VSNS_ENABLE | 244 tas2770->v_sense_slot); 245 if (ret) 246 goto end; 247 ret = snd_soc_component_update_bits(component, 248 TAS2770_TDM_CFG_REG6, 249 TAS2770_TDM_CFG_REG6_ISNS_MASK | 250 TAS2770_TDM_CFG_REG6_50_MASK, 251 TAS2770_TDM_CFG_REG6_ISNS_ENABLE | 252 tas2770->i_sense_slot); 253 254 end: 255 return ret; 256 } 257 258 static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) 259 { 260 int ret; 261 struct snd_soc_component *component = tas2770->component; 262 263 switch (samplerate) { 264 case 48000: 265 ret = snd_soc_component_update_bits(component, 266 TAS2770_TDM_CFG_REG0, 267 TAS2770_TDM_CFG_REG0_SMP_MASK, 268 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 269 if (ret) 270 goto end; 271 ret = snd_soc_component_update_bits(component, 272 TAS2770_TDM_CFG_REG0, 273 TAS2770_TDM_CFG_REG0_31_MASK, 274 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 275 if (ret) 276 goto end; 277 break; 278 case 44100: 279 ret = snd_soc_component_update_bits(component, 280 TAS2770_TDM_CFG_REG0, 281 TAS2770_TDM_CFG_REG0_SMP_MASK, 282 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 283 if (ret) 284 goto end; 285 ret = snd_soc_component_update_bits(component, 286 TAS2770_TDM_CFG_REG0, 287 TAS2770_TDM_CFG_REG0_31_MASK, 288 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 289 if (ret) 290 goto end; 291 break; 292 case 96000: 293 ret = snd_soc_component_update_bits(component, 294 TAS2770_TDM_CFG_REG0, 295 TAS2770_TDM_CFG_REG0_SMP_MASK, 296 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 297 if (ret) 298 goto end; 299 ret = snd_soc_component_update_bits(component, 300 TAS2770_TDM_CFG_REG0, 301 TAS2770_TDM_CFG_REG0_31_MASK, 302 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 303 break; 304 case 88200: 305 ret = snd_soc_component_update_bits(component, 306 TAS2770_TDM_CFG_REG0, 307 TAS2770_TDM_CFG_REG0_SMP_MASK, 308 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 309 if (ret) 310 goto end; 311 ret = snd_soc_component_update_bits(component, 312 TAS2770_TDM_CFG_REG0, 313 TAS2770_TDM_CFG_REG0_31_MASK, 314 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 315 break; 316 case 19200: 317 ret = snd_soc_component_update_bits(component, 318 TAS2770_TDM_CFG_REG0, 319 TAS2770_TDM_CFG_REG0_SMP_MASK, 320 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 321 if (ret) 322 goto end; 323 ret = snd_soc_component_update_bits(component, 324 TAS2770_TDM_CFG_REG0, 325 TAS2770_TDM_CFG_REG0_31_MASK, 326 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 327 if (ret) 328 goto end; 329 break; 330 case 17640: 331 ret = snd_soc_component_update_bits(component, 332 TAS2770_TDM_CFG_REG0, 333 TAS2770_TDM_CFG_REG0_SMP_MASK, 334 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 335 if (ret) 336 goto end; 337 ret = snd_soc_component_update_bits(component, 338 TAS2770_TDM_CFG_REG0, 339 TAS2770_TDM_CFG_REG0_31_MASK, 340 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 341 break; 342 default: 343 ret = -EINVAL; 344 } 345 346 end: 347 if (!ret) 348 tas2770->sampling_rate = samplerate; 349 return ret; 350 } 351 352 static int tas2770_hw_params(struct snd_pcm_substream *substream, 353 struct snd_pcm_hw_params *params, 354 struct snd_soc_dai *dai) 355 { 356 struct snd_soc_component *component = dai->component; 357 struct tas2770_priv *tas2770 = 358 snd_soc_component_get_drvdata(component); 359 int ret; 360 > 361 ret = tas2770_set_bitwidth(tas2770, params_format(params)); 362 if (ret < 0) 363 goto end; 364 365 366 ret = tas2770_set_samplerate(tas2770, params_rate(params)); 367 368 end: 369 return ret; 370 } 371 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-5.5 29/32] sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: e957204e732bc2916a241dc61dd7dd14e9a98350 [29/32] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout e957204e732bc2916a241dc61dd7dd14e9a98350 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:55: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:79: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +429 sound/core/pcm_dmaengine.c 371 372 /** 373 * snd_dmaengine_pcm_refine_runtime_hwparams - Refine runtime hw params 374 * @substream: PCM substream 375 * @dma_data: DAI DMA data 376 * @hw: PCM hw params 377 * @chan: DMA channel to use for data transfers 378 * 379 * Returns 0 on success, a negative error code otherwise. 380 * 381 * This function will query DMA capability, then refine the pcm hardware 382 * parameters. 383 */ 384 int snd_dmaengine_pcm_refine_runtime_hwparams( 385 struct snd_pcm_substream *substream, 386 struct snd_dmaengine_dai_dma_data *dma_data, 387 struct snd_pcm_hardware *hw, 388 struct dma_chan *chan) 389 { 390 struct dma_slave_caps dma_caps; 391 u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | 392 BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | 393 BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); 394 snd_pcm_format_t i; 395 int ret = 0; 396 397 if (!hw || !chan || !dma_data) 398 return -EINVAL; 399 400 ret = dma_get_slave_caps(chan, &dma_caps); 401 if (ret == 0) { 402 if (dma_caps.cmd_pause && dma_caps.cmd_resume) 403 hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; 404 if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) 405 hw->info |= SNDRV_PCM_INFO_BATCH; 406 407 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 408 addr_widths = dma_caps.dst_addr_widths; 409 else 410 addr_widths = dma_caps.src_addr_widths; 411 } 412 413 /* 414 * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep 415 * hw.formats set to 0, meaning no restrictions are in place. 416 * In this case it's the responsibility of the DAI driver to 417 * provide the supported format information. 418 */ 419 if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) 420 /* 421 * Prepare formats mask for valid/allowed sample types. If the 422 * dma does not have support for the given physical word size, 423 * it needs to be masked out so user space can not use the 424 * format which produces corrupted audio. 425 * In case the dma driver does not implement the slave_caps the 426 * default assumption is that it supports 1, 2 and 4 bytes 427 * widths. 428 */ > 429 for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { 430 int bits = snd_pcm_format_physical_width(i); 431 432 /* 433 * Enable only samples with DMA supported physical 434 * widths 435 */ 436 switch (bits) { 437 case 8: 438 case 16: 439 case 24: 440 case 32: 441 case 64: 442 if (addr_widths & (1 << (bits / 8))) 443 hw->formats |= pcm_format_to_bits(i); 444 break; 445 default: 446 /* Unsupported types */ 447 break; 448 } 449 } 450 451 return ret; 452 } 453 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams); 454 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-5.5 2/32] sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 9e28f6532c611c0c3fa759d2101aba9f0d41e860 [2/32] ASoC: fsl_mqs: Add MQS component driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 9e28f6532c611c0c3fa759d2101aba9f0d41e860 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer vim +182 sound/soc/fsl/fsl_mqs.c 178 179 static int fsl_mqs_probe(struct platform_device *pdev) 180 { 181 struct device_node *np = pdev->dev.of_node; > 182 struct device_node *gpr_np = 0; 183 struct fsl_mqs *mqs_priv; 184 void __iomem *regs; 185 int ret = 0; 186 187 mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); 188 if (!mqs_priv) 189 return -ENOMEM; 190 191 /* On i.MX6sx the MQS control register is in GPR domain 192 * But in i.MX8QM/i.MX8QXP the control register is moved 193 * to its own domain. 194 */ 195 if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) 196 mqs_priv->use_gpr = false; 197 else 198 mqs_priv->use_gpr = true; 199 200 if (mqs_priv->use_gpr) { 201 gpr_np = of_parse_phandle(np, "gpr", 0); 202 if (IS_ERR(gpr_np)) { 203 dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); 204 ret = PTR_ERR(gpr_np); 205 goto out; 206 } 207 208 mqs_priv->regmap = syscon_node_to_regmap(gpr_np); 209 if (IS_ERR(mqs_priv->regmap)) { 210 dev_err(&pdev->dev, "failed to get gpr regmap\n"); 211 ret = PTR_ERR(mqs_priv->regmap); 212 goto out; 213 } 214 } else { 215 regs = devm_platform_ioremap_resource(pdev, 0); 216 if (IS_ERR(regs)) 217 return PTR_ERR(regs); 218 219 mqs_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, 220 "core", 221 regs, 222 &fsl_mqs_regmap_config); 223 if (IS_ERR(mqs_priv->regmap)) { 224 dev_err(&pdev->dev, "failed to init regmap: %ld\n", 225 PTR_ERR(mqs_priv->regmap)); 226 return PTR_ERR(mqs_priv->regmap); 227 } 228 229 mqs_priv->ipg = devm_clk_get(&pdev->dev, "core"); 230 if (IS_ERR(mqs_priv->ipg)) { 231 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 232 PTR_ERR(mqs_priv->ipg)); 233 goto out; 234 } 235 } 236 237 mqs_priv->mclk = devm_clk_get(&pdev->dev, "mclk"); 238 if (IS_ERR(mqs_priv->mclk)) { 239 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 240 PTR_ERR(mqs_priv->mclk)); 241 goto out; 242 } 243 244 dev_set_drvdata(&pdev->dev, mqs_priv); 245 pm_runtime_enable(&pdev->dev); 246 247 return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, 248 &fsl_mqs_dai, 1); 249 out: 250 if (!IS_ERR(gpr_np)) 251 of_node_put(gpr_np); 252 253 return ret; 254 } 255 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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