June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/4] ASoC: wm_adsp: Fix event for preloader
by Charles Keepax 21 Jun '22

21 Jun '22

The preloader controls on ADSP should return a value of 1 if the preloader value was changed, update to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6d7fd88243aa8..a7784ac15dde6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -997,7 +997,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); } - return 0; + return 1; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); -- 2.30.2

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[PATCH] ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error
by shumingf＠realtek.com 21 Jun '22

21 Jun '22

From: Shuming Fan <shumingf(a)realtek.com> The initial settings will be written before the codec probe function. But, the rt711->component doesn't be assigned yet. If IO error happened during initial settings operations, it will cause the kernel panic. This patch changed component->dev to slave->dev to fix this issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 9d226b1cb7e9..925268121901 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -34,7 +34,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to set private value: %06x <= %04x ret=%d\n", addr, value, ret); @@ -50,7 +50,7 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to get private value: %06x => %04x ret=%d\n", addr, *value, ret); -- 2.36.1

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[PATCH 1/4] ASoC: fsl-asoc-card: add WM8904 support
by Alifer Moraes 21 Jun '22

21 Jun '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> The codec WM8904 can use internal FLL as PLL source. Whenever the PLL source is not an external MCLK, this source must be setup during hw_params callback otherwise the BCLK could be wrongly evaluated. The SND_SOC_BIAS_PREPARE event is raised after the hw_params callback, so there is no need to set again PLL and SYSCLK and actually there's no need at all the set_bias_level function. Also, when esai is used, a dedicated snd_soc_dai_set_tdm_slot call is required. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/fsl/fsl-asoc-card.c | 47 +++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ee945505281..817dbc1ec635 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "imx-audmux.h" #include "../codecs/sgtl5000.h" +#include "../codecs/wm8904.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" @@ -257,6 +258,38 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) } } + if (of_device_is_compatible(dev->of_node, "fsl,imx-audio-wm8904")) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0, 0, 2, + params_physical_width(params)); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->pll_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + return 0; } @@ -651,6 +684,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->card.set_bias_level = NULL; + priv->codec_priv.mclk_id = WM8904_CLK_FLL; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + } } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -900,6 +946,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8904", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, -- 2.25.1

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Intel HDMI probe regression on IVB (and older?)
by Takashi Iwai 21 Jun '22

21 Jun '22

Hi, we've got a regression report about Intel HDMI. It seems that the recent change to skip the component binding (commit c9db8a30d9f0) throws away the devices incorrectly on IvyBridge. I guess the similar issue could happen on older chips. The bug report is found at https://bugzilla.opensuse.org/show_bug.cgi?id=1200611 Judging from the logs there, on IvyBridge, the GPU is 00:02.0 0380: 8086:0162 (rev 09) DeviceName: Onboard IGD Subsystem: 1043:84ca Flags: bus master, fast devsel, latency 0, IRQ 27 Memory at f7800000 (64-bit, non-prefetchable) [size=4M] Memory at d0000000 (64-bit, prefetchable) [size=256M] I/O ports at f000 [size=64] Capabilities: <access denied> Kernel driver in use: i915 Kernel modules: i915 while HD-audio is 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 1043:84a8 Flags: bus master, fast devsel, latency 0, IRQ 31 Memory at f7d10000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel Unlike Haswell, IVB has no dedicated HDMI audio controller, and it's connected on HD-audio bus of 00:1b.0. Kai, could you check this issue? thanks, Takashi

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[PATCH] ALSA: hda: Fix discovery of i915 graphics PCI device
by Takashi Iwai 21 Jun '22

21 Jun '22

It's been reported that the recent fix for skipping the component-binding with D-GPU caused a regression on some systems; it resulted in the completely missing component binding with i915 GPU. The problem was the use of pci_get_class() function. It matches with the full PCI class bits, while we want to match only partially the PCI base class bits. So, when a system has an i915 graphics device with the PCI class 0380, it won't hit because we're looking for only the PCI class 0300. This patch fixes i915_gfx_present() to look up each PCI device and match with PCI base class explicitly instead of pci_get_class(). Fixes: c9db8a30d9f0 ("ALSA: hda/i915 - skip acomp init if no matching display") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Tested-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.opensuse.org/show_bug.cgi?id=1200611 Link: https://lore.kernel.org/r/87bkunztec.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 3f35972e1cf7..161a9711cd63 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -119,21 +119,18 @@ static int i915_component_master_match(struct device *dev, int subcomponent, /* check whether Intel graphics is present and reachable */ static int i915_gfx_present(struct pci_dev *hdac_pci) { - unsigned int class = PCI_BASE_CLASS_DISPLAY << 16; struct pci_dev *display_dev = NULL; - bool match = false; - do { - display_dev = pci_get_class(class, display_dev); - - if (display_dev && display_dev->vendor == PCI_VENDOR_ID_INTEL && + for_each_pci_dev(display_dev) { + if (display_dev->vendor == PCI_VENDOR_ID_INTEL && + (display_dev->class >> 16) == PCI_BASE_CLASS_DISPLAY && connectivity_check(display_dev, hdac_pci)) { pci_dev_put(display_dev); - match = true; + return true; } - } while (!match && display_dev); + } - return match; + return false; } /** -- 2.35.3

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[PATCH 0/3] ASoC: sun50i-codec-analog: Internal bias support
by Samuel Holland 21 Jun '22

21 Jun '22

This series adds support for enabling the codec's internal microphone bias, which is needed on at least some versions of the PinePhone. Arnaud Ferraris (2): ASoC: dt-bindings: sun50i-codec: Add binding for internal bias ASoC: sun50i-codec-analog: Add support for internal bias Samuel Holland (1): arm64: dts: allwinner: pinephone: Enable internal HMIC bias .../allwinner,sun50i-a64-codec-analog.yaml | 5 ++++ .../allwinner/sun50i-a64-pinephone-1.0.dts | 4 +++ .../allwinner/sun50i-a64-pinephone-1.1.dts | 4 +++ sound/soc/sunxi/sun50i-codec-analog.c | 29 +++++++++++++++++++ 4 files changed, 42 insertions(+) -- 2.35.1

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[PATCH 00/10] timer: rename tstamp methods with the word real_time
by GitHub pull_request - opened 21 Jun '22

21 Jun '22

alsa-project/alsa-gobject pull request #78 was opened from takaswie: In ALSATimer library some methods are named with the word `tstamp`. On the other hand, in ALSASeq library the similar methods are named with the word `real_time`. For consistency, this patchset renames methods in ALSATimer library. As a result, the word `tstamp` is not used at all. ``` Takashi Sakamoto (10): utils: successfully parse the string teminated by '\n' timer: move ALSATimer.TickEvent files for future rename timer: move ALSATimer.TstampEvent files for future rename timer: tick-time-event: rename boxed structure timer: real-time-event: rename boxed structure timer: enums: rename to ALSATimer.EventType.TICK_TIME and REAL_TIME timer: enums: rename to ALSATimer.RealTimeEventType timer: instance-status: rename method with real time timer: query: rename to ALSATimer.get_real_time_clock_id() seq: event: fix description about methods relevant to time stamp samples/timer | 26 +++---- src/seq/event.c | 16 ++-- src/timer/alsatimer-enum-types.h | 74 +++++++++---------- src/timer/alsatimer.h | 4 +- src/timer/alsatimer.map | 20 ++--- src/timer/instance-params.c | 20 ++--- src/timer/instance-params.h | 4 +- src/timer/instance-status.c | 22 +++--- src/timer/instance-status.h | 3 +- src/timer/meson.build | 8 +- src/timer/query.c | 16 ++-- src/timer/query.h | 2 +- src/timer/real-time-event.c | 65 ++++++++++++++++ src/timer/real-time-event.h | 25 +++++++ src/timer/tick-event.c | 51 ------------- src/timer/tick-event.h | 23 ------ src/timer/tick-time-event.c | 50 +++++++++++++ src/timer/tick-time-event.h | 22 ++++++ src/timer/tstamp-event.c | 64 ---------------- src/timer/tstamp-event.h | 24 ------ src/timer/user-instance.c | 56 +++++++------- src/timer/user-instance.h | 17 +++-- src/utils/string.c | 2 +- tests/alsatimer-enums | 8 +- tests/alsatimer-functions | 2 +- tests/alsatimer-instance-status | 2 +- ...tstamp-event => alsatimer-real-time-event} | 4 +- ...r-tick-event => alsatimer-tick-time-event} | 4 +- tests/alsatimer-user-instance | 8 +- tests/meson.build | 4 +- 30 files changed, 324 insertions(+), 322 deletions(-) create mode 100644 src/timer/real-time-event.c create mode 100644 src/timer/real-time-event.h delete mode 100644 src/timer/tick-event.c delete mode 100644 src/timer/tick-event.h create mode 100644 src/timer/tick-time-event.c create mode 100644 src/timer/tick-time-event.h delete mode 100644 src/timer/tstamp-event.c delete mode 100644 src/timer/tstamp-event.h rename tests/{alsatimer-tstamp-event => alsatimer-real-time-event} (84%) rename tests/{alsatimer-tick-event => alsatimer-tick-time-event} (83%) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/78 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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GoXLR Broken in 1.2.7.1
by GitHub issues - opened 21 Jun '22

21 Jun '22

alsa-project/alsa-ucm-conf issue #175 was opened from FrostyCoolSlug: The alsa profiles no longer set correctly for the GoXLR or GoXLR Mini since 5dcd69. >From what I can tell with some quick tests, if the DirectCardName isn't set, the profile doesn't seem to apply anymore. Removing the direct config, or adding a DirectCardName seems to solve the issue (although obviously, neither are ideal). Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/175 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v5 00/14] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 21 Jun '22

21 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. changes since v4: - Fully remove tlv remnants from control add apis - Remove unnecessary debug - Rename variable to be more generic - Remove redundent length check from read/write control apis - Use SNDRV_CTL_ELEM_IFACE_CARD for firmware load controls - Make kcontrol add/remove synchronous - Load firmware asynchronous when loading via control - Used cached controls when reloading firmware; only delete controls when removing the driver itself - Improve kcontrol remove - Fix control write + notify - Cleanup of unnecessary code - Fix race condition when loading firmware before playback - Ensure errors are properly propogated - Fix include for Module parameters Stefan Binding (13): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 4 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 902 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 39 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 254 +++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 39 ++ sound/pci/hda/patch_realtek.c | 27 +- 12 files changed, 1277 insertions(+), 5 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

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Enabling fsl_sai MCLK before codec probe
by Michael Welling 20 Jun '22

20 Jun '22

So I have a tricky race condition that comes up with hardware that combines imx8 with a pcm1681 codec. The pcm1681 requires that MCLK be enabled for 65536 cycles before accessing the I2C management port. The imx8 sai driver on the other hand does not start the built in MCLK until a confluence of events that is not triggered until just before streaming starts which is too late. Is there a clean way to start the imx8 sai MCLK before the codec probe occurs? Or have it started and delayed during the codec probe before it tries to access i2c? We have a working solution but it is a bit overreaching and adds a callback function to the snd_soc_dai_ops struct to start the clock, and adds the code to the fsl_sai.c to toggle the appropriate bits in the SoC sai driver to make a clock appear and registers to the new function callback. Then the simple-card was amended to add the appropriate enabling through the new callback such that the clock is on long enough before the codec is probed. There was some suggestion of caching the i2c registers until the clock is enabled but this would be difficult given the clock is not started until streaming starts and we don't want to stream until the codec is initialized. I am looking for a solution that is minimal and robust. Any pointers would be greatly appreciated.

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