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[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 3

[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 122 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

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[PATCH] ALSA: usb-audio: Add input gain and master output mixer controls for RME Babyface Pro
by Stefan Stistrup 06 Aug '24

06 Aug '24

Add missing input gain and master output mixer controls for RME Babyface Pro This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup <sstistrup(a)gmail.com> --- sound/usb/mixer_quirks.c | 162 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 161 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d771..26298e06b838 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,113 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + } else { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + } + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2906,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2938,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2997,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) { + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + } else { + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + } + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, -- 2.45.2

2 2

[RFC PATCH 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Aug '24

06 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. Shengjiu Wang (6): ALSA: compress: add Sample Rate Converter codec support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 + include/uapi/sound/compress_params.h | 9 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 769 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 10 files changed, 1284 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH v1] ASoC: meson: axg-fifo: set option to use raw spinlock
by Arseniy Krasnov 06 Aug '24

06 Aug '24

Raw spinlock is needed here, because with enabled PREEMPT_RT, spinlock_t become preemptible, but this regmap lock could be acquired in IRQ handler. Found by lockdep: [ ] ============================= [ ] [ BUG: Invalid wait context ] [ ] 6.9.9-sdkernel #1 Tainted: G O [ ] ----------------------------- [ ] aplay/413 is trying to lock: [ ] ffff000003930018 (axg_fifo:356:(&axg_fifo_regmap_cfg)->lock){....}-{3:3},c [ ] other info that might help us debug this: [ ] context-{2:2} [ ] no locks held by aplay/413. [ ] stack backtrace: [ ] CPU: 0 PID: 413 Comm: aplay Tainted: G O 6.9.9-kernel #1 [ ] Hardware name: SberDevices SberBoom Mini (DT) [ ] Call trace: [ ] dump_backtrace+0x98/0xf0 [ ] show_stack+0x18/0x24 [ ] dump_stack_lvl+0x90/0xd0 [ ] dump_stack+0x18/0x24 [ ] __lock_acquire+0x9dc/0x1f10 [ ] lock_acquire.part.0+0xe8/0x228 [ ] lock_acquire+0x68/0x84 [ ] _raw_spin_lock_irqsave+0x60/0x88 [ ] regmap_lock_spinlock+0x18/0x2c [ ] regmap_read+0x3c/0x78 [ ] axg_fifo_pcm_irq_block+0x4c/0xc8 [ ] __handle_irq_event_percpu+0xa4/0x2f8 [ ] handle_irq_event+0x4c/0xbc [ ] handle_fasteoi_irq+0xa4/0x23c [ ] generic_handle_domain_irq+0x2c/0x44 [ ] gic_handle_irq+0x40/0xc4 [ ] call_on_irq_stack+0x24/0x4c [ ] do_interrupt_handler+0x80/0x84 [ ] el0_interrupt+0x5c/0x124 [ ] __el0_irq_handler_common+0x18/0x24 [ ] el0t_32_irq_handler+0x10/0x1c [ ] el0t_32_irq+0x194/0x198 Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-fifo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index ecb3eb7a9723d..a22298f74b35a 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -328,6 +328,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .val_bits = 32, .reg_stride = 4, .max_register = FIFO_CTRL2, + .use_raw_spinlock = true, }; int axg_fifo_probe(struct platform_device *pdev) -- 2.30.1

4 12

[PATCH] ASoC: codecs: lpass-macro: fix missing codec version
by Johan Hovold 05 Aug '24

05 Aug '24

Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- Cc: regressions(a)lists.linux.dev #regzbot introduced: 378918d59181 sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) -- 2.44.2

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Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 05 Aug '24

05 Aug '24

Hi, > > You get twice more capacity based on that configuration. I can't answer the > > second question because not working with field. But both of that configurations > > are used by customers. Adding Neal if he wants to add something more to it. > > Just to add a comment as I work directly with our customers. The main reason > this support is important is for our older SoCs, zynq and zynqmp. > > Most of our customers are using QSPI flash as the first boot memory to get > from the boot ROM to u-boot. They then typically use other memories, such as > eMMC for the Linux kernel, OS and file system. Agreed and that's probably the most prominent use case for NOR flashes anyway. > The issue we have on the zynq and zynqmp SoCs is that the boot ROM (code that > cannot be changed) will not boot from an OSPI flash. It will only boot from a > QSPI flash. This is what is forcing many of our customers down the QSPI path. > Since many of these customers are interested in additional speed and memory > size, they then end up using a parallel or stacked configuration because they > cannot use an OSPI with zynq or zynqmp. Above you've said, the bootloader is stored on the NOR flash and the bulk storage is eMMC. So why do you need bigger NOR flashes (where even the biggest NOR flash isn't enough)? I also don't understand "the boot ROM will just boot from QSPI". First, you cannot connect an octal flash anyway, because you only have an QSPI controller, right? Secondly, normally the bootrom will (at least) boot the first stage using normal (single line) SPI commands. Is that not the case for zynq and zynqmp? > All of our newer SoCs can boot from OSPI. I agree with you that if someone > could choose OSPI for performance, they would, so I do not expect parallel or > stacked configurations with our newer SoCs. Ok, but then the argument with bigger flashes are void, because you are back to be bound to one OSPI flash. > I get why you see this configuration as a niche, but for us, it is a very > large niche because zynq and zynqmp are two of our most successful SoC > families. Fair enough. But please find a way to support it without butchering the whole core. -michael

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Re: [PATCH v2] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 17:50:44 +0200, Stefan Binding wrote: > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls. > > These controls were never intended to be exposed to users but the > firmware doesn't mark them hidden, so make the driver ignore them. > > Any changes in the coefficients handled by these controls needs to > be matched to the individual system by SSID, which is already handled > using the tuning file, when firmware is loaded, so UCM should not be > setting these controls anyway. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Applied to for-linus branch now. Thanks. Takashi

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Re: [PATCH v3] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 16:31:39 +0200, Simon Trimmer wrote: > > A number of laptops have gone to market with old firmware versions that > export controls that have since been hidden, but we can't just install a > newer firmware because the firmware for each product is customized and > qualified by the OEM. The issue is that alsactl save and restore has no > idea what controls are good to persist which can lead to > misconfiguration. > > There is no reason that the UCM or user should need to interact with any > of the ALSA controls for the firmware coefficients so they can be > removed entirely, this also simplifies the driver. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > Changes in v3: > - Remove ability to add firmware coefficients as ALSA controls > entirely > > Changes in v2: > - v1 was accidentally the backport version for older kernels instead of > the one for v6.11 Applied now to for-linus branch. Thanks. Takashi

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ucm2: sof-soundwire: add rt712 VA device
by GitHub pull_request - opened 05 Aug '24

05 Aug '24

alsa-project/alsa-ucm-conf pull request #435 was opened from shumingfan: Add support headphone/headset mic/speaker/dmic for rt712 VA Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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