October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/5] ASoC: SOF: Intel: add flags to turn on SSP clocks early
by Pierre-Louis Bossart 05 Oct '21

05 Oct '21

With the chip shortage, some GeminiLake Intel-based designs were respun and now rely on codecs that need the SSP bit clock turned on in the hw_params stage, not the trigger stage. This patchset mirrors the flags added in the SOF DAI_CONFIG IPC, and sets the flags when this capability is indicated as necessary in the topology files where the SSP configuration is stored. We initially considered a more generic solution with an on-demand SSP clock activation using the common clock framework. This would be a more elegant solution indeed, but it would have required more intrusive changes that would conflict with the SOF multi-client support (in-development), and more backport hassles on product branches. The on-demand activation of clocks is still a desired feature that will be enabled at a later point. Bard Liao (1): ASoC: SOF: dai-intel: add SOF_DAI_INTEL_SSP_CLKCTRL_MCLK/BCLK_ES bits Pierre-Louis Bossart (4): ASoC: SOF: dai: mirror group_id definition added in firmware ASoC: SOF: dai: include new flags for DAI_CONFIG ASoC: SOF: Intel: hda: add new flags for DAI_CONFIG ASoC: SOF: Intel: hda-dai: improve SSP DAI handling for dynamic pipelines include/sound/sof/dai-intel.h | 4 ++ include/sound/sof/dai.h | 10 ++++- sound/soc/sof/intel/hda-dai.c | 82 ++++++++++++++++++++++++++++++++++- sound/soc/sof/intel/hda.c | 6 +++ sound/soc/sof/sof-audio.c | 4 ++ 5 files changed, 103 insertions(+), 3 deletions(-) -- 2.25.1

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[PATCH] ASoC: amd: acp-rt5645: Constify static snd_soc_ops
by Rikard Falkeborn 05 Oct '21

05 Oct '21

The struct cz_aif1_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/amd/acp-rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..6d5c547a32de 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -91,7 +91,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_ops cz_aif1_ops = { +static const struct snd_soc_ops cz_aif1_ops = { .hw_params = cz_aif1_hw_params, }; -- 2.33.0

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[PATCH] ASoC: SOF: OF: Add fw_path and tplg_path parameters
by Daniel Baluta 05 Oct '21

05 Oct '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This allows specifying an alternate path for SOF firmware or SOF topology. This is particularly useful for i.MX when running Linux vs Android. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-of-dev.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index f0f819a46456..885430a42226 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -7,12 +7,21 @@ #include <linux/firmware.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/pm_runtime.h> #include <sound/sof.h> #include "ops.h" #include "imx/imx-ops.h" +static char *fw_path; +module_param(fw_path, charp, 0444); +MODULE_PARM_DESC(fw_path, "alternate path for SOF firmware."); + +static char *tplg_path; +module_param(tplg_path, charp, 0444); +MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology."); + /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) static struct sof_dev_desc sof_of_imx8qxp_desc = { @@ -87,9 +96,15 @@ static int sof_of_probe(struct platform_device *pdev) sof_pdata->dev = &pdev->dev; sof_pdata->fw_filename = desc->default_fw_filename; - /* TODO: read alternate fw and tplg filenames from DT */ - sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path; - sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path; + if (fw_path) + sof_pdata->fw_filename_prefix = fw_path; + else + sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path; + + if (tplg_path) + sof_pdata->tplg_filename_prefix = tplg_path; + else + sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path; /* set callback to be called on successful device probe to enable runtime_pm */ sof_pdata->sof_probe_complete = sof_of_probe_complete; -- 2.27.0

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[PATCH AUTOSEL 5.14 15/40] ASoC: SOF: loader: release_firmware() on load failure to avoid batching
by Sasha Levin 05 Oct '21

05 Oct '21

From: Marc Herbert <marc.herbert(a)intel.com> [ Upstream commit 8a8e1813ffc35111fc0b6db49968ceb0e1615ced ] Invoke release_firmware() when the firmware fails to boot in sof_probe_continue(). The request_firmware() framework must be informed of failures in sof_probe_continue() otherwise its internal "batching" feature (different from caching) cached the firmware image forever. Attempts to correct the file in /lib/firmware/ were then silently and confusingly ignored until the next reboot. Unloading the drivers did not help because from their disconnected perspective the firmware had failed so there was nothing to release. Also leverage the new snd_sof_fw_unload() function to simplify the snd_sof_device_remove() function. Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20210916085008.28929-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 4 +--- sound/soc/sof/loader.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e4dd4a86363..59d0d7b2b55c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev) dev_warn(dev, "error: %d failed to prepare DSP for device removal", ret); - snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); @@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); /* release firmware */ - release_firmware(pdata->fw); - pdata->fw = NULL; + snd_sof_fw_unload(sdev); return 0; } diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..9c3f251a0dd0 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware); void snd_sof_fw_unload(struct snd_sof_dev *sdev) { /* TODO: support module unloading at runtime */ + release_firmware(sdev->pdata->fw); + sdev->pdata->fw = NULL; } EXPORT_SYMBOL(snd_sof_fw_unload); -- 2.33.0

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[PATCH AUTOSEL 5.14 08/40] ALSA: usb-audio: Unify mixer resume and reset_resume procedure
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 7b9cf9036609428e845dc300aec13822ba2c4ab3 ] USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu <enshuo(a)chromium.org> Cc: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsM… Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 6abfc9d079e7..fa75b7e72ad1 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1020,7 +1020,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1046,7 +1046,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1066,20 +1066,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1101,7 +1091,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9b713b4a5ec4..fa7cf982d39e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3655,33 +3655,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3702,7 +3685,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ea41e7a1f7bf..16567912b998 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -122,7 +121,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7d00..4a4d3361ac04 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); -- 2.33.0

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[PATCH AUTOSEL 5.14 05/40] ALSA: oxfw: fix transmission method for Loud models based on OXFW971
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 64794d6db49730d22f440aef0cf4da98a56a4ea3 ] Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210913021042.10085-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } -- 2.33.0

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[PATCH AUTOSEL 5.14 03/40] ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic
by Sasha Levin 05 Oct '21

05 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 58eafe1ff52ee1ce255759fc15729519af180cbb ] The SoundWire BEs make use of 'stream' functions for .prepare and .trigger. These functions will in turn force a Bank Switch, which implies a wait operation. Mark SoundWire BEs as nonatomic for consistency, but keep all other types of BEs as is. The initialization of .nonatomic is done outside of the create_sdw_dailink helper to avoid adding more parameters to deal with a single exception to the rule that BEs are atomic. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20210907184436.33152-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1a867c73a48e..cb3afc4519cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -860,6 +860,11 @@ static int create_sdw_dailink(struct device *dev, int *be_index, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(link, dai_links + (*be_index)++, playback, group_id); -- 2.33.0

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Re: [PATCH v2 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:41:31AM +0800, cy_huang wrote: > Very appoligized. I forget to migrate this line from my test 5.10 kernel. > Without this line will cause sound card codec link fail. > Please use the patch v2 to review the bindings and source code. Sorry, saw v1 before I saw this but I guess the issues still apply - a quick check seems like it.

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Re: [PATCH 2/2] ASoC: rt9120: Add rt9210 audio amplifier support
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:55AM +0800, cy_huang wrote: > +static const char * const sdo_select_text[] = { > + "NONE", "INTF", "FINAL", "RMS Detect" > +}; Why not None and Final? > + if (event == SND_SOC_DAPM_PRE_PMU) > + snd_soc_component_write(comp, RT9120_REG_ERRRPT, 0); > + else if (event == SND_SOC_DAPM_POST_PMU) > + msleep(RT9120_AMPON_WAITMS); > + else if (event == SND_SOC_DAPM_POST_PMD) > + msleep(RT9120_AMPOFF_WAITMS); This should be a switch statement, it'd be clearer. > + /* Default config volume to 0dB */ > + snd_soc_component_write(comp, RT9120_REG_MSVOL, 0x180); > + /* Mute by default */ > + snd_soc_component_update_bits(comp, RT9120_REG_VOLRAMP, > + RT9120_MUTE_MASK, RT9120_MUTE_MASK); As ever you should leave the defaults at whatever the hardware defaults to, the defaults for one machine may not be suitable for another so we shouldn't be trying to do that in software.

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Re: [PATCH 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:54AM +0800, cy_huang wrote: > + richtek,use-dvdd-1p8v: > + description: Indicate DVDD 1P8V is used, default for 3P3V or 5V design > + type: boolean I would expect this to be done through the regulator bindings, they would allow the driver to query the supply voltage.

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