September 2020 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH 03/17] drm/exynos: stop setting DMA_ATTR_NON_CONSISTENT
by Christoph Hellwig 15 Sep '20

15 Sep '20

On Mon, Sep 14, 2020 at 06:34:02PM +0300, Sergei Shtylyov wrote: > On 9/14/20 5:44 PM, Christoph Hellwig wrote: > > > DMA_ATTR_NON_CONSISTENT is a no-op except on PARISC and some mips > > configs, so don't set it in this ARM specific driver. > > Hm, PARICS and ARM capitalized but mips in lower case? :-) I guess it should also be PA-RISC if we start nitpicking..

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Possible bug for ubuntu 20.04 with CONFIG_SND_MAX_CARDS=256
by Olivier SCHMITT 15 Sep '20

15 Sep '20

Hello dear all, I have been compiling kernels for a long time with CONFIG_SND_MAX_CARDS = 256 and increasing the sample rate. I switched to ubuntu 20.04 server lts. Everything is working fine until I compile alsa-lib. git clone git: //git.alsa-project.org/alsa-lib.git alsa-lib ./gitcompile --with-max-cards = 256 But after the make install, the plughw, dmix and dnsoop disappear. Do you have an idea? I uploaded alsa-info on: http://alsa-project.org/db/?f=8860b8bbb5abd5a84d3002717d68206db7da4a9c Thanks in advance.

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[PATCH 1/2] leds: trigger: audio: Add audio jack plugging indicator led
by Hui Wang 15 Sep '20

15 Sep '20

On the Lenovo P520 front panel, there is a headset plugging indicator led. According to LENOVO's requirement, this led should be on if the front headset audio jack is plugged, otherwise the led should be off. We tested with windows, if we plugged a headphone/headset in that jack, the led was on, if we removed the headphone/headset from that jack, the led was off. Here we add JACKPLUG led in the ledtrig_audio, then we could call ledtrig_audio_set() in the alsa driver to control that led. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- drivers/leds/trigger/Kconfig | 4 ++-- drivers/leds/trigger/ledtrig-audio.c | 7 +++++-- include/linux/leds.h | 1 + 3 files changed, 8 insertions(+), 4 deletions(-) diff --git a/drivers/leds/trigger/Kconfig b/drivers/leds/trigger/Kconfig index ce9429ca6dde..a61893a6e3e4 100644 --- a/drivers/leds/trigger/Kconfig +++ b/drivers/leds/trigger/Kconfig @@ -138,10 +138,10 @@ config LEDS_TRIGGER_PATTERN If unsure, say N config LEDS_TRIGGER_AUDIO - tristate "Audio Mute LED Trigger" + tristate "Audio Mute and Jack Plugging LED Trigger" help This allows LEDs to be controlled by audio drivers for following - the audio mute and mic-mute changes. + the audio mute, mic-mute and jack plugging changes. If unsure, say N endif # LEDS_TRIGGERS diff --git a/drivers/leds/trigger/ledtrig-audio.c b/drivers/leds/trigger/ledtrig-audio.c index f76621e88482..fa66bb982400 100644 --- a/drivers/leds/trigger/ledtrig-audio.c +++ b/drivers/leds/trigger/ledtrig-audio.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// Audio Mute LED trigger +// Audio Mute And Jack Plugging LED trigger // #include <linux/kernel.h> @@ -29,6 +29,8 @@ static int __init ledtrig_audio_init(void) &ledtrig_audio[LED_AUDIO_MUTE]); led_trigger_register_simple("audio-micmute", &ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_register_simple("audio-jackplug", + &ledtrig_audio[LED_AUDIO_JACKPLUG]); return 0; } module_init(ledtrig_audio_init); @@ -37,8 +39,9 @@ static void __exit ledtrig_audio_exit(void) { led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MUTE]); led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_JACKPLUG]); } module_exit(ledtrig_audio_exit); -MODULE_DESCRIPTION("LED trigger for audio mute control"); +MODULE_DESCRIPTION("LED trigger for audio mute and jack plugging control"); MODULE_LICENSE("GPL v2"); diff --git a/include/linux/leds.h b/include/linux/leds.h index 6a8d6409c993..1858c2d30ecd 100644 --- a/include/linux/leds.h +++ b/include/linux/leds.h @@ -584,6 +584,7 @@ struct led_pattern { enum led_audio { LED_AUDIO_MUTE, /* master mute LED */ LED_AUDIO_MICMUTE, /* mic mute LED */ + LED_AUDIO_JACKPLUG, /* audio jack plug indicator LED */ NUM_AUDIO_LEDS }; -- 2.17.1

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increase CONFIG_SND_MAX_CARDS and avalable sample rate
by Olivier SCHMITT 14 Sep '20

14 Sep '20

Hello dear all, It's possible de increase CONFIG_SND_MAX_CARDS to 256 and increse sample rate to 384khz. Actualy i build my own kernel as: sed -i 's/SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,/SNDRV_PCM_RATE_CONTINUOUS,/g' ./sound/drivers/aloop.c sed -i 's/192000/768000/g' ./sound/drivers/aloop.c grep -rnl './' -e "CONFIG_SND_MAX_CARDS=32" | xargs -i@ sed -i 's/CONFIG_SND_MAX_CARDS=32/CONFIG_SND_MAX_CARDS=256/g' @ search and replace 32 by 256 in and ./debian.master/config/annotations:CONFIG_SND_MAX_CARDS and for alsa utils: git clone git://git.alsa-project.org/alsa-lib.git alsa-lib ./gitcompile --with-max-cards=256 make install git clone git://git.alsa-project.org/alsa-utils.git alsa-utils ./gitcompile --with-max-cards=256 make install Thanks in advance.

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[PATCH v2 00/11] USB: new USB control message helper functions
by Greg Kroah-Hartman 14 Sep '20

14 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h Changes from v1: - added acks from Takashi Iwai - dropped changes to one function in patch 04 thanks to review from Alan Stern - typo fix in comment in patch 01 - added new patch 11 to remove some unneeded checks in the sound drivers for endpoint statuses that would always be true. *** BLURB HERE *** Greg Kroah-Hartman (11): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() ALSA: remove calls to usb_pipe_type_check for control endpoints drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 99 ++++++++---------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 31 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 335 insertions(+), 342 deletions(-) -- 2.28.0

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Re: [PATCH] ASoC: tas6424: disable regulators when probe error
by Mark Brown 14 Sep '20

14 Sep '20

On Mon, Sep 14, 2020 at 11:52:15PM +0900, Chanho Park wrote: > On Mon, Sep 14, 2020 at 8:27 PM Mark Brown <broonie(a)kernel.org> wrote: > > Please think hard before including complete backtraces in upstream > > reports, they are very large and contain almost no useful information > > relative to their size so often obscure the relevant content in your > > message. If part of the backtrace is usefully illustrative (it often is > > for search engines if nothing else) then it's usually better to pull out > > the relevant sections. > Thanks for your suggestion. > I'll respin this patch without the call stack messages. It's fine, I've already applied it - just please bear this in mind for future things.

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[PATCH] ALSA: rockchip_i2s: fix a possible divide-by-zero bug in rockchip_i2s_hw_params()
by Tuo Li 14 Sep '20

14 Sep '20

The variable bclk_rate is checked in: if (bclk_rate && mclk_rate % bclk_rate) This indicates that bclk_rate can be zero. If so, a divide-by-zero bug will occur: div_bclk = mclk_rate / bclk_rate; To fix this possible bug, the function returns -EINVAL when bclk_rate is zero. Signed-off-by: Tuo Li <tuoli96(a)outlook.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 d1438753edb4..593299675b8c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -279,7 +279,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, if (i2s->is_master_mode) { mclk_rate = clk_get_rate(i2s->mclk); bclk_rate = 2 * 32 * params_rate(params); - if (bclk_rate && mclk_rate % bclk_rate) + if (bclk_rate == 0 || mclk_rate % bclk_rate) return -EINVAL; div_bclk = mclk_rate / bclk_rate; -- 2.17.1

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[PATCH 0/4] ASoC: SOF: DSP core management fixes for 5.10
by Kai Vehmanen 14 Sep '20

14 Sep '20

This series contains some improvements to how DSP core management is done in SOF, and adds a distinction between cores managed by the host versus cores managed by the DSP. Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda-loader: s/master/primary Ranjani Sridharan (3): ASoC: SOF: rename cores_mask to host_managed_cores_mask ASoC: SOF: Intel: hda: modify core_power_up/down op ASoC: SOF: Intel: remove the HDA_DSP_CORE_MASK() macro sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 6 +++--- sound/soc/sof/intel/cnl.c | 15 ++++----------- sound/soc/sof/intel/hda-dsp.c | 20 +++++++++++++++++--- sound/soc/sof/intel/hda-loader.c | 11 +++++------ sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 3 --- sound/soc/sof/intel/shim.h | 2 +- sound/soc/sof/intel/tgl.c | 2 +- 10 files changed, 34 insertions(+), 31 deletions(-) -- 2.27.0

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[PATCH] ASoC: SOF: Intel: Use DMI oem string search for tgl_max98373_rt5682
by Kai Vehmanen 14 Sep '20

14 Sep '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> DMI product name is used to support system variants based out of tgl_max98373_rt5682 in current implementation. Replace this DMI search with DMI_OEM_STRING. Coreboot(BIOS used in these systems) is setting the needed DMI_OEM_STRING field to uniquely identify these systems. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/sof/sof-pci-dev.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 9a6f10ede427..ddbb9fe7cc06 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -123,7 +123,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { .callback = sof_rt5682_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Volteer"), - DMI_MATCH(DMI_PRODUCT_NAME, "Terrador"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index ec62d9337d68..3ec380945466 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -50,7 +50,7 @@ static const struct dmi_system_id sof_tplg_table[] = { .callback = sof_tplg_cb, .matches = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Volteer"), - DMI_MATCH(DMI_PRODUCT_NAME, "Terrador"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = "sof-tgl-rt5682-ssp0-max98373-ssp2.tplg", }, -- 2.27.0

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[PATCH 0/2] ASoC: mediatek: mt8183-da7219: support machine driver for rt1015p
by Tzung-Bi Shih 14 Sep '20

14 Sep '20

The series reuses mt8183-da7219-max98357.c for supporting machine driver with rt1015p speaker amplifier. The 1st patch adds document for the new proposed compatible string. The 2nd patch changes the machine driver to support "RT1015P" codec. Tzung-Bi Shih (2): ASoC: dt-bindings: mt8183-da7219: add compatible string for using rt1015p ASoC: mediatek: mt8183-da7219: support machine driver with rt1015p .../bindings/sound/mt8183-da7219-max98357.txt | 1 + sound/soc/mediatek/Kconfig | 1 + .../mediatek/mt8183/mt8183-da7219-max98357.c | 40 +++++++++++++++++++ 3 files changed, 42 insertions(+) -- 2.28.0.526.ge36021eeef-goog

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[PATCH 1/2] ASoC: rt1015: Fix DC calibration on bypass boost mode
by derek.fang＠realtek.com 14 Sep '20

14 Sep '20

From: "derek.fang" <derek.fang(a)realtek.com> Fix the DC calibration unsuccessful issue on rt1015 bypass boost mode. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt1015.c | 48 +++++++++++++++++++++++++++++++++-------------- sound/soc/codecs/rt1015.h | 1 + 2 files changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 548f686..ba1b6b3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -484,6 +484,33 @@ static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } +static void rt1015_calibrate(struct rt1015_priv *rt1015) +{ + struct snd_soc_component *component = rt1015->component; + struct regmap *regmap = rt1015->regmap; + + snd_soc_dapm_mutex_lock(&component->dapm); + regcache_cache_bypass(regmap, true); + + regmap_write(regmap, RT1015_PWR1, 0xd7df); + regmap_write(regmap, RT1015_PWR4, 0x00b2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); + msleep(500); + regmap_write(regmap, RT1015_PWR1, 0x0); + + regcache_cache_bypass(regmap, false); + regcache_mark_dirty(regmap); + regcache_sync(regmap); + snd_soc_dapm_mutex_unlock(&component->dapm); +} + static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -494,20 +521,12 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, if (!rt1015->dac_is_used) { rt1015->bypass_boost = ucontrol->value.integer.value[0]; - if (rt1015->bypass_boost == RT1015_Bypass_Boost) { - snd_soc_component_write(component, - RT1015_PWR4, 0x00b2); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL8, 0x2008); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL9, 0x0140); - snd_soc_component_write(component, - RT1015_GAT_BOOST, 0x0efe); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000d); - msleep(500); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000e); + if (rt1015->bypass_boost == RT1015_Bypass_Boost && + !rt1015->cali_done) { + rt1015_calibrate(rt1015); + rt1015->cali_done = 1; + + regmap_write(rt1015->regmap, RT1015_MONO_DYNA_CTRL, 0x0010); } } else dev_err(component->dev, "DAC is being used!\n"); @@ -888,6 +907,7 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; + rt1015->cali_done = 0; snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); return 0; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 7bd159e..4d11f58 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -389,6 +389,7 @@ struct rt1015_priv { int bypass_boost; int amp_ver; int dac_is_used; + int cali_done; }; #endif /* __RT1015_H__ */ -- 2.7.4

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[PATCH v2 0/2] ASoC: ti: j721e-evm: Support for j7200 variant
by Peter Ujfalusi 14 Sep '20

14 Sep '20

Hi, Changes since v1: - Suffix the 2359296000 constant with 'u' to silence C90 warning When j7200 SOM is connected to the CPB, the audio setup is a bit different: Only 48KHz family have clock path, 44.1KHz is not supported. Update the binding documentation and add support for the j7200 version of CPB to the driver. Regards, Peter --- Peter Ujfalusi (2): ASoC: dt-bindings: ti,j721e-cpb-audio: Document support for j7200-cpb ASoC: ti: j721e-evm: Add support for j7200-cpb audio .../bindings/sound/ti,j721e-cpb-audio.yaml | 92 ++++++++++++++----- sound/soc/ti/j721e-evm.c | 11 +++ 2 files changed, 81 insertions(+), 22 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH] ASoC: stm32: sai: add pm_runtime support
by Olivier Moysan 14 Sep '20

14 Sep '20

Enable support of pm_runtime on STM32 SAI driver to allow SAI power state monitoring. pm_runtime_put_autosuspend() is called from ASoC framework on pcm device close. The pmdown_time delay is available in runtime context, and may be set in SAI driver to take into account shutdown delay on playback. However, this shutdown delay is already handled in the DAPMs of the audio codec linked to SAI CPU DAI. So, the choice is made, not to support this delay on CPU DAI side. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3fb9513cedb2..3aa1cf262402 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/asoundef.h> @@ -1559,10 +1560,14 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) + if (ret) { snd_dmaengine_pcm_unregister(&pdev->dev); + return ret; + } - return ret; + pm_runtime_enable(&pdev->dev); + + return 0; } static int stm32_sai_sub_remove(struct platform_device *pdev) @@ -1572,6 +1577,7 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) clk_unprepare(sai->pdata->pclk); snd_dmaengine_pcm_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); return 0; } -- 2.17.1

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ucm2: USB-Audio: Add CaptureMixerElem for Lenovo ThinkStation P620
by GitHub pull_request - opened 14 Sep '20

14 Sep '20

alsa-project/alsa-ucm-conf pull request #51 was opened from khfeng: The Switch and Volume mixer start to work after device firmware update. Add MixerElem to support them. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/51 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/51.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] ASoC: ak4458: Add DSD support for ak4458 and ak4497
by Shengjiu Wang 14 Sep '20

14 Sep '20

ak4458 can't support DSD512 format, but ak4497 can, so add a new variable (dsd512) in ak4458_drvdata to distinguish these two platforms. Add a new kcontrol for ak4497 codec for ak4497 has a specific pin selection. In hw_params(), calculate bit clock according to different DSD format and configure DSD registers. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add more commit description. sound/soc/codecs/ak4458.c | 82 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/ak4458.h | 5 ++- 2 files changed, 79 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 763e6839428f..2ef8b591eb79 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -31,11 +31,13 @@ static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; + bool dsd512; /* DSD512 is supported or not */ }; /* AK4458 Codec Private Data */ struct ak4458_priv { struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; + const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpiod; @@ -136,6 +138,10 @@ static const char * const ak4458_ats_select_texts[] = { /* DIF2 bit Audio Interface Format Setting(BICK fs) */ static const char * const ak4458_dif_select_texts[] = {"32fs,48fs", "64fs",}; +/* DSD input pin select */ +static const char * const ak4497_dsd_input_pin_select[] = { + "16_17_19pin", "3_4_5pin"}; + static const struct soc_enum ak4458_dac1_dem_enum = SOC_ENUM_SINGLE(AK4458_01_CONTROL2, 1, ARRAY_SIZE(ak4458_dem_select_texts), @@ -175,6 +181,10 @@ static const struct soc_enum ak4458_dif_enum = SOC_ENUM_SINGLE(AK4458_00_CONTROL1, 3, ARRAY_SIZE(ak4458_dif_select_texts), ak4458_dif_select_texts); +static const struct soc_enum ak4497_dsdp_enum = + SOC_ENUM_SINGLE(AK4458_09_DSD2, 2, + ARRAY_SIZE(ak4497_dsd_input_pin_select), + ak4497_dsd_input_pin_select); static int get_digfil(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -282,6 +292,7 @@ static const struct snd_kcontrol_new ak4497_snd_controls[] = { SOC_ENUM("AK4497 Sound Mode", ak4458_sm_enum), SOC_ENUM("AK4497 Attenuation transition Time Setting", ak4458_ats_enum), + SOC_ENUM("AK4497 DSD Data Input Pin", ak4497_dsdp_enum), }; /* ak4497 dapm widgets */ @@ -325,12 +336,54 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - int nfs1; - u8 format; + u8 format, dsdsel0, dsdsel1; + int nfs1, dsd_bclk; nfs1 = params_rate(params); ak4458->fs = nfs1; + /* calculate bit clock */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + dsd_bclk = nfs1 * params_physical_width(params); + switch (dsd_bclk) { + case 2822400: + dsdsel0 = 0; + dsdsel1 = 0; + break; + case 5644800: + dsdsel0 = 1; + dsdsel1 = 0; + break; + case 11289600: + dsdsel0 = 0; + dsdsel1 = 1; + break; + case 22579200: + if (ak4458->drvdata->dsd512) { + dsdsel0 = 1; + dsdsel1 = 1; + } else { + dev_err(dai->dev, "DSD512 not supported.\n"); + return -EINVAL; + } + break; + default: + dev_err(dai->dev, "Unsupported dsd bclk.\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AK4458_06_DSD1, + AK4458_DSDSEL_MASK, dsdsel0); + snd_soc_component_update_bits(component, AK4458_09_DSD2, + AK4458_DSDSEL_MASK, dsdsel1); + break; + } + /* Master Clock Frequency Auto Setting Mode Enable */ snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); @@ -355,6 +408,9 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_DSP_B: format = AK4458_DIF_32BIT_MSB; break; + case SND_SOC_DAIFMT_PDM: + format = AK4458_DIF_32BIT_MSB; + break; default: return -EINVAL; } @@ -393,6 +449,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_PDM: ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; break; default: @@ -401,6 +458,12 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + /* DSD mode */ + snd_soc_component_update_bits(component, AK4458_02_CONTROL3, + AK4458_DP_MASK, + ak4458->fmt == SND_SOC_DAIFMT_PDM ? + AK4458_DP_MASK : 0); + ak4458_rstn_control(component, 0); ak4458_rstn_control(component, 1); @@ -472,7 +535,10 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, #define AK4458_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_DSD_U8 |\ + SNDRV_PCM_FMTBIT_DSD_U16_LE |\ + SNDRV_PCM_FMTBIT_DSD_U32_LE) static const unsigned int ak4458_rates[] = { 8000, 11025, 16000, 22050, @@ -668,11 +734,13 @@ static const struct regmap_config ak4458_regmap = { static const struct ak4458_drvdata ak4458_drvdata = { .dai_drv = &ak4458_dai, .comp_drv = &soc_codec_dev_ak4458, + .dsd512 = false, }; static const struct ak4458_drvdata ak4497_drvdata = { .dai_drv = &ak4497_dai, .comp_drv = &soc_codec_dev_ak4497, + .dsd512 = true, }; static const struct dev_pm_ops ak4458_pm = { @@ -684,7 +752,6 @@ static const struct dev_pm_ops ak4458_pm = { static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; - const struct ak4458_drvdata *drvdata; int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); @@ -698,7 +765,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, ak4458); ak4458->dev = &i2c->dev; - drvdata = of_device_get_match_data(&i2c->dev); + ak4458->drvdata = of_device_get_match_data(&i2c->dev); ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", GPIOD_OUT_LOW); @@ -720,8 +787,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, - drvdata->dai_drv, 1); + ret = devm_snd_soc_register_component(ak4458->dev, + ak4458->drvdata->comp_drv, + ak4458->drvdata->dai_drv, 1); if (ret < 0) { dev_err(ak4458->dev, "Failed to register CODEC: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index f906215f7e4e..9548c5d78621 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -83,4 +83,7 @@ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) -#endif /* _AK4458_H */ +#define AK4458_DSDSEL_MASK (0x1 << 0) +#define AK4458_DP_MASK (0x1 << 7) + +#endif -- 2.27.0

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Re: [PATCH] ASoC: tas6424: disable regulators when probe error
by Mark Brown 14 Sep '20

14 Sep '20

On Sat, Sep 12, 2020 at 11:31:51AM +0900, Chanho Park wrote: > To fix up below errors when tas6424 chip is not available, this patch > disables regulators before returning error. > > [ 9.367234] tas6424 2-006a: unable to reset device: -6 > [ 9.499118] Call trace: > [ 9.501550] _regulator_put.part.21+0x158/0x160 > [ 9.506065] regulator_put+0x34/0x48 > [ 9.509624] regulator_bulk_free+0x30/0x50 > [ 9.513705] devm_regulator_bulk_release+0x18/0x20 > [ 9.518483] release_nodes+0x1b8/0x2a0 > [ 9.522215] devres_release_all+0x34/0x50 Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections.

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control: snd_ctl_open(): which names are acceptable?
by Tanjeff Moos 13 Sep '20

13 Sep '20

Hello Alsa developers, I'm currently enhancing the libalsa API documentation. In the process, I try to understand how each API function works and what the underlying concepts are. The snd_ctl_open() call takes a name, which is a C string (const char*). For my sound card, the names "hw:CARD=PCH" and as well as "hw:2" (2 is the card index) worked. Which strings are acceptable as name? Are there maybe even more formats? Is there some documentation on this? Regards, Tanjeff

1 0

[GIT PULL] ASoC fixes for v5.9-rc4
by Mark Brown 12 Sep '20

12 Sep '20

The following changes since commit d012a7190fc1fd72ed48911e77ca97ba4521bccd: Linux 5.9-rc2 (2020-08-23 14:08:43 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.9-rc4 for you to fetch changes up to 1a5ce48fd667128e369fdc7fb87e21539aed21b5: ASoC: tlv320adcx140: Wake up codec before accessing register (2020-09-08 17:26:03 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.9 Most of this is various driver specific fixes, none of which are terribly exciting in themselves, plus one core fix adding and using a new DAI lookup function to deal with a lockdep warning. ---------------------------------------------------------------- Brent Lu (1): ASoC: intel: atom: Add period size constraint Camel Guo (2): ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev ASoC: tlv320adcx140: Wake up codec before accessing register Cezary Rojewski (2): ASoC: Intel: haswell: Fix power transition refactor ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure Dinghao Liu (1): ASoC: qcom: common: Fix refcount imbalance on error Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1 Jerome Brunet (1): ASoC: meson: axg-toddr: fix channel order on g12 platforms Kuninori Morimoto (3): ASoC: pcm3168a: ignore 0 Hz settings ASoC: ti: fixup ams_delta_mute() function name ASoC: soc-core: add snd_soc_find_dai_with_mutex() Mark Brown (2): Merge tag 'v5.9-rc2' into asoc-5.9 Merge series "ASoC: Fix return check for devm_regmap_init_sdw()" from Vinod Koul <vkoul(a)kernel.org>: Mateusz Gorski (1): ASoC: Intel: skl_hda_dsp_generic: Fix NULLptr dereference in autosuspend delay Rander Wang (1): ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case Stephan Gerhold (1): ASoC: qcom: Set card->owner to avoid warnings Sylwester Nawrocki (2): ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811 ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions Vinod Koul (5): ASoC: max98373: Fix return check for devm_regmap_init_sdw() ASoC: rt1308-sdw: Fix return check for devm_regmap_init_sdw() ASoC: rt711: Fix return check for devm_regmap_init_sdw() ASoC: rt715: Fix return check for devm_regmap_init_sdw() ASoC: rt700: Fix return check for devm_regmap_init_sdw() include/sound/soc.h | 4 + sound/soc/codecs/max98373-sdw.c | 4 +- sound/soc/codecs/pcm3168a.c | 7 + sound/soc/codecs/rt1308-sdw.c | 4 +- sound/soc/codecs/rt700-sdw.c | 4 +- sound/soc/codecs/rt711-sdw.c | 4 +- sound/soc/codecs/rt715-sdw.c | 4 +- sound/soc/codecs/tlv320adcx140.c | 28 ++-- sound/soc/codecs/wm8994.c | 10 ++ sound/soc/codecs/wm_hubs.c | 3 + sound/soc/codecs/wm_hubs.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 11 ++ sound/soc/intel/boards/bytcr_rt5640.c | 10 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 7 +- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++++--------------- sound/soc/meson/axg-toddr.c | 24 +++- sound/soc/qcom/apq8016_sbc.c | 1 + sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/common.c | 6 +- sound/soc/qcom/sdm845.c | 1 + sound/soc/qcom/storm.c | 1 + sound/soc/soc-core.c | 13 ++ sound/soc/soc-dai.c | 4 +- sound/soc/soc-pcm.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- 26 files changed, 206 insertions(+), 139 deletions(-)

2 1

[PATCH v10 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

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

1 3

[PATCH v9 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name defined in sc7180-lpass.h. https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 267 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 422 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.618.gf4bc123cb7-goog

3 7

[PATCH v2] ALSA: firewire: Replace tasklet with work
by Takashi Iwai 11 Sep '20

11 Sep '20

The tasklet is an old API that should be deprecated, usually can be converted to another decent API. In FireWire driver, a tasklet is still used for offloading the AMDTP PCM stream handling. It can be achieved gracefully with a work queued, too. This patch replaces the tasklet usage in firewire-lib driver with a simple work. The conversion is fairly straightforward but for the in_interrupt() checks that are replaced with the check using the current_work(). Note that in_interrupt() in amdtp_packet tracepoint is still kept as is. This is the place that is probed by both softirq of 1394 OHCI and a user task of a PCM application, and the work handling is already filtered in amdtp_domain_stream_pcm_pointer(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Drop in_interrupt() conversion in tracepoint sound/firewire/amdtp-stream.c | 25 +++++++++++++------------ sound/firewire/amdtp-stream.h | 2 +- 2 files changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ee1c428b1fd3..4e2f2bb7879f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -64,7 +64,7 @@ #define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header. #define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing. -static void pcm_period_tasklet(struct tasklet_struct *t); +static void pcm_period_work(struct work_struct *work); /** * amdtp_stream_init - initialize an AMDTP stream structure @@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); - tasklet_setup(&s->period_tasklet, pcm_period_tasklet); + INIT_WORK(&s->period_work, pcm_period_work); s->packet_index = 0; init_waitqueue_head(&s->callback_wait); @@ -203,7 +203,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, // Linux driver for 1394 OHCI controller voluntarily flushes isoc // context when total size of accumulated context header reaches - // PAGE_SIZE. This kicks tasklet for the isoc context and brings + // PAGE_SIZE. This kicks work for the isoc context and brings // callback in the middle of scheduled interrupts. // Although AMDTP streams in the same domain use the same events per // IRQ, use the largest size of context header between IT/IR contexts. @@ -333,7 +333,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload); */ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) { - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; } @@ -437,13 +437,14 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - tasklet_hi_schedule(&s->period_tasklet); + queue_work(system_highpri_wq, &s->period_work); } } -static void pcm_period_tasklet(struct tasklet_struct *t) +static void pcm_period_work(struct work_struct *work) { - struct amdtp_stream *s = from_tasklet(s, t, period_tasklet); + struct amdtp_stream *s = container_of(work, struct amdtp_stream, + period_work); struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); if (pcm) @@ -794,7 +795,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (current_work() == &s->period_work) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1184,7 +1185,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // This function is called in software IRQ context of - // period_tasklet or process context. + // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. @@ -1195,9 +1196,9 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software - // IRQ context of the period_tasklet. Then, no need to flush the + // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above - if (!in_interrupt()) { + if (current_work() != &s->period_work) { // Queued packet should be processed without any kernel // preemption to keep latency against bus cycle. preempt_disable(); @@ -1263,7 +1264,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s) return; } - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 703b710aaf7f..2ceb57d1d58e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -163,7 +163,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; - struct tasklet_struct period_tasklet; + struct work_struct period_work; snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; -- 2.16.4

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[PATCH 1/4] ASoC: odroid: Add missing properties
by Krzysztof Kozlowski 11 Sep '20

11 Sep '20

Usage of "unevaluatedProperties: false" is not correct as it suppresses warnings about all undocumented properties. Instead, add all missing properties. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Fixup for: https://lore.kernel.org/lkml/30b8ea43-c2d0-5ddf-dc16-a8fe80d47c38@samsung.c… --- .../devicetree/bindings/sound/samsung,odroid.yaml | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index de1be3d6d1e9..e8122bc87362 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -28,6 +28,11 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: The user-visible name of this sound complex. + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + clocks: true + cpu: type: object properties: @@ -69,7 +74,7 @@ required: - cpu - codec -unevaluatedProperties: false +additionalProperties: false examples: - | -- 2.17.1

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[PATCH] ASoC: core: Optionaly add the component driver name to the component name
by Dafna Hirschfeld 11 Sep '20

11 Sep '20

The ASoC platform driver for Mediatek MT8173 registers several components with identical name. This cause an error: "Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present!" when adding debugfs directory for those components. To fix this, the function 'fmt_single_name' can receive the name of the component driver and concatenate it to the name of the debugfs directory for the component. Signed-off-by: Dafna Hirschfeld <dafna.hirschfeld(a)collabora.com> --- sound/soc/soc-core.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 74df22486e30..f470e713b030 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2227,7 +2227,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, int *id, const char *component_driver_name) { const char *devname = dev_name(dev); char *found, *name; @@ -2263,6 +2263,12 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (component_driver_name) { + char *tmp = name; + + name = devm_kasprintf(dev, GFP_KERNEL, "%s-%s", name, component_driver_name); + devm_kfree(dev, tmp); + } return name; } @@ -2327,7 +2333,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, &dai->id, NULL); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2460,7 +2466,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, &component->id, driver->name); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.17.1

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[PATCH] ASoC: ak4458: Add DSD support for ak4458 and ak4497
by Shengjiu Wang 11 Sep '20

11 Sep '20

ak4458 can't support DSD512 format, but ak4497 can, so add a new variable in ak4458_drvdata to distinguish these two platform. In hw_params(), calculate bit clock according to different DSD format and configure DSD register. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 82 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/ak4458.h | 5 ++- 2 files changed, 79 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 763e6839428f..9054964d5cfa 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -31,11 +31,13 @@ static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; + bool dsd512; /* DSD512 is supported or not */ }; /* AK4458 Codec Private Data */ struct ak4458_priv { struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; + const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpiod; @@ -136,6 +138,10 @@ static const char * const ak4458_ats_select_texts[] = { /* DIF2 bit Audio Interface Format Setting(BICK fs) */ static const char * const ak4458_dif_select_texts[] = {"32fs,48fs", "64fs",}; +/* DSD input pin select */ +static const char * const ak4497_dsd_input_path_select[] = { + "16_17_19pin", "3_4_5pin"}; + static const struct soc_enum ak4458_dac1_dem_enum = SOC_ENUM_SINGLE(AK4458_01_CONTROL2, 1, ARRAY_SIZE(ak4458_dem_select_texts), @@ -175,6 +181,10 @@ static const struct soc_enum ak4458_dif_enum = SOC_ENUM_SINGLE(AK4458_00_CONTROL1, 3, ARRAY_SIZE(ak4458_dif_select_texts), ak4458_dif_select_texts); +static const struct soc_enum ak4497_dsdp_enum = + SOC_ENUM_SINGLE(AK4458_09_DSD2, 2, + ARRAY_SIZE(ak4497_dsd_input_path_select), + ak4497_dsd_input_path_select); static int get_digfil(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -282,6 +292,7 @@ static const struct snd_kcontrol_new ak4497_snd_controls[] = { SOC_ENUM("AK4497 Sound Mode", ak4458_sm_enum), SOC_ENUM("AK4497 Attenuation transition Time Setting", ak4458_ats_enum), + SOC_ENUM("AK4497 DSD Data Input Pin", ak4497_dsdp_enum), }; /* ak4497 dapm widgets */ @@ -325,12 +336,54 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - int nfs1; - u8 format; + u8 format, dsdsel0, dsdsel1; + int nfs1, dsd_bclk; nfs1 = params_rate(params); ak4458->fs = nfs1; + /* calculate bit clock */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + dsd_bclk = nfs1 * params_physical_width(params); + switch (dsd_bclk) { + case 2822400: + dsdsel0 = 0; + dsdsel1 = 0; + break; + case 5644800: + dsdsel0 = 1; + dsdsel1 = 0; + break; + case 11289600: + dsdsel0 = 0; + dsdsel1 = 1; + break; + case 22579200: + if (ak4458->drvdata->dsd512) { + dsdsel0 = 1; + dsdsel1 = 1; + } else { + dev_err(dai->dev, "DSD512 not supported.\n"); + return -EINVAL; + } + break; + default: + dev_err(dai->dev, "Unsupported dsd bclk.\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AK4458_06_DSD1, + AK4458_DSDSEL_MASK, dsdsel0); + snd_soc_component_update_bits(component, AK4458_09_DSD2, + AK4458_DSDSEL_MASK, dsdsel1); + break; + } + /* Master Clock Frequency Auto Setting Mode Enable */ snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); @@ -355,6 +408,9 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_DSP_B: format = AK4458_DIF_32BIT_MSB; break; + case SND_SOC_DAIFMT_PDM: + format = AK4458_DIF_32BIT_MSB; + break; default: return -EINVAL; } @@ -393,6 +449,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_PDM: ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; break; default: @@ -401,6 +458,12 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + /* DSD mode */ + snd_soc_component_update_bits(component, AK4458_02_CONTROL3, + AK4458_DP_MASK, + ak4458->fmt == SND_SOC_DAIFMT_PDM ? + AK4458_DP_MASK : 0); + ak4458_rstn_control(component, 0); ak4458_rstn_control(component, 1); @@ -472,7 +535,10 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, #define AK4458_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_DSD_U8 |\ + SNDRV_PCM_FMTBIT_DSD_U16_LE |\ + SNDRV_PCM_FMTBIT_DSD_U32_LE) static const unsigned int ak4458_rates[] = { 8000, 11025, 16000, 22050, @@ -668,11 +734,13 @@ static const struct regmap_config ak4458_regmap = { static const struct ak4458_drvdata ak4458_drvdata = { .dai_drv = &ak4458_dai, .comp_drv = &soc_codec_dev_ak4458, + .dsd512 = false, }; static const struct ak4458_drvdata ak4497_drvdata = { .dai_drv = &ak4497_dai, .comp_drv = &soc_codec_dev_ak4497, + .dsd512 = true, }; static const struct dev_pm_ops ak4458_pm = { @@ -684,7 +752,6 @@ static const struct dev_pm_ops ak4458_pm = { static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; - const struct ak4458_drvdata *drvdata; int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); @@ -698,7 +765,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, ak4458); ak4458->dev = &i2c->dev; - drvdata = of_device_get_match_data(&i2c->dev); + ak4458->drvdata = of_device_get_match_data(&i2c->dev); ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", GPIOD_OUT_LOW); @@ -720,8 +787,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, - drvdata->dai_drv, 1); + ret = devm_snd_soc_register_component(ak4458->dev, + ak4458->drvdata->comp_drv, + ak4458->drvdata->dai_drv, 1); if (ret < 0) { dev_err(ak4458->dev, "Failed to register CODEC: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index f906215f7e4e..9548c5d78621 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -83,4 +83,7 @@ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) -#endif /* _AK4458_H */ +#define AK4458_DSDSEL_MASK (0x1 << 0) +#define AK4458_DP_MASK (0x1 << 7) + +#endif -- 2.27.0

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[PATCH v8 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name defined in sc7180-lpass.h. https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 270 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 425 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.526.ge36021eeef-goog

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