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[PATCH] ASoC: wm8524: remove constraint for FE-BE
by Fabio Estevam 22 Apr '22

22 Apr '22

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> The constraint is not needed for back end bistream since the sample rate is fixed by dts and the constraint is propagated to front end bistream as they share the same snd_soc_pcm_runtime. Fixes: 007b6a54c305 ("ASoC: codecs: add wm8524 codec driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Viorel Suman <viorel.suman(a)nxp.com> Tested-by: Fabio Estevam <festevam(a)denx.de> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/codecs/wm8524.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs/wm8524.c') diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 81f858f..442a598 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -61,6 +61,7 @@ static int wm8524_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct wm8524_priv *wm8524 = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = substream->private_data; /* The set of sample rates that can be supported depends on the * MCLK supplied to the CODEC - enforce this. @@ -71,9 +72,10 @@ static int wm8524_startup(struct snd_pcm_substream *substream, return -EINVAL; } - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &wm8524->rate_constraint); + if (!rtd->dai_link->be_hw_params_fixup) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8524->rate_constraint); gpiod_set_value_cansleep(wm8524->mute, 1); -- cgit v1.1

2 2

Re: [RFC PATCH 0/5] Apple Macs machine-level ASoC driver
by Mark Brown 22 Apr '22

22 Apr '22

On Thu, Mar 31, 2022 at 05:04:32PM +0200, Martin Povišer wrote: > > On 31. 3. 2022, at 16:18, Mark Brown <broonie(a)kernel.org> wrote: > > Yes, having two devices driving the bus at the same time wouldn't be > > great. How is the TDM slot selection for the signals done in the > > hardware, I'm not seeing anything immediately obvious in the driver? > > I'd have thought that things would be implemented such that you could > > implement speaker protection on all speakers simultaneously but perhaps > > not. > I don’t know. I would have to go study the details of this. Should I see > if I can find a combination of ‘ASI1 Sel’ ‘VSENSE’ ‘ISENSE’ settings > that would lead to driver conflict on one of the models, or is there > a chance we could hide those controls just on the basis of ‘it doesn’t > do anything usable and is possibly dangerous’? If ISENSE and VSENSE output are controlled by the same mux as routing then we should lock one of the controls out for at least stereo devices (it might be a good idea to check if the output is actually high Z when ISENSE and VSENSE are off rather than just driving zeros, if not it definitely has to be the routing control). My instinct is that it's better to preserve the ability to implement speaker protection in future since that is something that'd be broadly useful, especially if someone comes up with a generic speaker protection implementation in which case there should be an awful lot of systems out there which could benefit. > >> That’s the reasoning anyway. To reiterate, seems to me the controls > >> are useless/confusing at best and dangerous at worst. > > I'm just not seeing an issue for the slot selection. > Yeah, agreed there’s no (damage) issue as we should to proper volume > caps anyway. Though see above about how ISENSE/VSENSE output slot is controlled I guess :/

2 11

[PATCH] ASoC: Intel: avs: Enable AVS driver only on x86 platforms
by Amadeusz Sławiński 22 Apr '22

22 Apr '22

Only supported platform for AVS are x86 machines, so there is no reason for it to build on other architectures. Fixes: 274d79e51875 ("ASoC: Intel: avs: Configure modules according to their type") Reported-by: kernel test robot <lkp(a)intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c364ddf22267..5dc075673b97 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -211,7 +211,7 @@ config SND_SOC_INTEL_KEEMBAY config SND_SOC_INTEL_AVS tristate "Intel AVS driver" - depends on PCI && ACPI + depends on X86 && PCI && ACPI depends on COMMON_CLK select SND_SOC_ACPI select SND_SOC_TOPOLOGY -- 2.25.1

2 1

[PATCH] ASoC: rt9120: Correct the reg 0x09 size to one byte
by cy_huang 22 Apr '22

22 Apr '22

From: ChiYuan Huang <cy_huang(a)richtek.com> Correct the reg 0x09 size to one byte. Signed-off-by: ChiYuan Huang <cy_huang(a)richtek.com> --- Sorry, in the original datasheet, reg 0x09 size is marked as 2 bytes. But actually it's one byte RG size. --- sound/soc/codecs/rt9120.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index 7aa1772..6e0d7cf 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -341,7 +341,6 @@ static int rt9120_get_reg_size(unsigned int reg) { switch (reg) { case 0x00: - case 0x09: case 0x20 ... 0x27: return 2; case 0x30 ... 0x3D: -- 2.7.4

2 1

[PATCH] ALSA: usb-audio: Always apply the hw constraints for implicit fb sync
by Takashi Iwai 22 Apr '22

22 Apr '22

Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync"), we apply the hw constraints for the implicit feedback sync to make the secondary open aligned with the already opened stream setup. This change assumed that the secondary open is performed after the first stream has been already set up, and adds the hw constraints to sync with the first stream's parameters only when the EP setup for the first stream was confirmed at the open time. However, most of applications handling the full-duplex operations do open both playback and capture streams at first, then set up both streams. This results in skipping the additional hw constraints since the counter-part stream hasn't been set up yet at the open of the second stream, and it eventually leads to "incompatible EP" error in the end. This patch corrects the behavior by always applying the hw constraints for the implicit fb sync. The hw constraint rules are defined so that they check the sync EP dynamically at each invocation, instead. This covers the concurrent stream setups better and lets the hw refine calls resolving to the right configuration. Also this patch corrects a minor error that has existed in the debug print that isn't built as default. Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This patched was developed primarily as an attemp for the regression of Pioneer devices. Although Pioneer devices have still some issues after this patch, this one itself should be good for all devices in general. Keith, Dylan, could you verify that this doesn't break things for your devices? sound/usb/pcm.c | 171 ++++++++++++++++++++++++++++++------------------ 1 file changed, 108 insertions(+), 63 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 56079901769f..f71965bf815f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -663,7 +663,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, check_fmts.bits[1] = (u32)(fp->formats >> 32); snd_mask_intersect(&check_fmts, fmts); if (snd_mask_empty(&check_fmts)) { - hwc_debug(" > check: no supported format %d\n", fp->format); + hwc_debug(" > check: no supported format 0x%llx\n", fp->formats); return 0; } /* check the channels */ @@ -775,24 +775,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, rmin, rmax); } -static int hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int apply_hw_params_format_bits(struct snd_mask *fmt, u64 fbits) { - struct snd_usb_substream *subs = rule->private; - const struct audioformat *fp; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - u64 fbits; u32 oldbits[2]; int changed; - hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); - fbits = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!hw_check_valid_format(subs, params, fp)) - continue; - fbits |= fp->formats; - } - oldbits[0] = fmt->bits[0]; oldbits[1] = fmt->bits[1]; fmt->bits[0] &= (u32)fbits; @@ -806,6 +793,24 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } +static int hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct audioformat *fp; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u64 fbits; + + hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); + fbits = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + fbits |= fp->formats; + } + return apply_hw_params_format_bits(fmt, fbits); +} + static int hw_rule_period_time(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -833,64 +838,92 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return apply_hw_params_minmax(it, pmin, UINT_MAX); } -/* apply PCM hw constraints from the concurrent sync EP */ -static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) +/* get the EP or the sync EP for implicit fb when it's already set up */ +static const struct snd_usb_endpoint * +get_sync_ep_from_substream(struct snd_usb_substream *subs) { struct snd_usb_audio *chip = subs->stream->chip; - struct snd_usb_endpoint *ep; const struct audioformat *fp; - int err; + const struct snd_usb_endpoint *ep; list_for_each_entry(fp, &subs->fmt_list, list) { ep = snd_usb_get_endpoint(chip, fp->endpoint); if (ep && ep->cur_rate) - goto found; + return ep; if (!fp->implicit_fb) continue; /* for the implicit fb, check the sync ep as well */ ep = snd_usb_get_endpoint(chip, fp->sync_ep); if (ep && ep->cur_rate) - goto found; + return ep; } - return 0; + return NULL; +} + +/* additional hw constraints for implicit feedback mode */ +static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - found: - if (!find_format(&subs->fmt_list, ep->cur_format, ep->cur_rate, - ep->cur_channels, false, NULL)) { - usb_audio_dbg(chip, "EP 0x%x being used, but not applicable\n", - ep->ep_num); + ep = get_sync_ep_from_substream(subs); + if (!ep) return 0; - } - usb_audio_dbg(chip, "EP 0x%x being used, using fixed params:\n", - ep->ep_num); - usb_audio_dbg(chip, "rate=%d, period_size=%d, periods=%d\n", - ep->cur_rate, ep->cur_period_frames, - ep->cur_buffer_periods); + hwc_debug("applying %s\n", __func__); + return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format)); +} - runtime->hw.formats = subs->formats; - runtime->hw.rate_min = runtime->hw.rate_max = ep->cur_rate; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - runtime->hw.periods_min = runtime->hw.periods_max = - ep->cur_buffer_periods; +static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1); - if (err < 0) - return err; + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; - err = snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - ep->cur_period_frames, - ep->cur_period_frames); - if (err < 0) - return err; + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate); +} + +static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; - return 1; /* notify the finding */ + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + return apply_hw_params_minmax(it, ep->cur_period_frames, + ep->cur_period_frames); +} + +static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + const struct snd_usb_endpoint *ep; + struct snd_interval *it; + + ep = get_sync_ep_from_substream(subs); + if (!ep) + return 0; + + hwc_debug("applying %s\n", __func__); + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS); + return apply_hw_params_minmax(it, ep->cur_buffer_periods, + ep->cur_buffer_periods); } /* @@ -899,20 +932,11 @@ static int apply_hw_constraint_from_sync(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { - struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed = -1; int err; - mutex_lock(&chip->mutex); - err = apply_hw_constraint_from_sync(runtime, subs); - mutex_unlock(&chip->mutex); - if (err < 0) - return err; - if (err > 0) /* found the matching? */ - goto add_extra_rules; - runtime->hw.formats = subs->formats; runtime->hw.rate_min = 0x7fffffff; @@ -964,7 +988,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; -add_extra_rules: err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, @@ -993,6 +1016,28 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; } + /* additional hw constraints for implicit fb */ + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + hw_rule_period_size_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + hw_rule_periods_implicit_fb, subs, + SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + return 0; } -- 2.26.2

2 1

[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs support for HP Laptops
by Andy Chi 22 Apr '22

22 Apr '22

On HP Laptops, requires the same ALC285_FIXUP_HP_GPIO_LED quirk to make its audio LEDs work. So apply the quirk, and make it the last one since it's an LED quirk. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0cba2f19a772..29de8d7959c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9023,12 +9023,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8981, "HP Elite Dragonfly G3", ALC245_FIXUP_CS35L41_SPI_4), SND_PCI_QUIRK(0x103c, 0x898e, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x898f, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Fix mute led issue on thinkpad with cs35l41 s-codec
by Hui Wang 22 Apr '22

22 Apr '22

The quirk ALC287_FIXUP_CS35L41_I2C_2 needs to chain the quirk ALC269_FIXUP_THINKPAD_ACPI, otherwise the mute led will not work if a thinkpad machine applies that quirk. And it will be safe if non-thinkpad machines apply that quirk since hda_fixup_thinkpad_acpi() will check and return in this case. Fixes: ae7abe36e352e ("ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c0c593f3c0a..f9c3b2c9ca12 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8769,6 +8769,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC287_FIXUP_CS35L41_I2C_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, -- 2.25.1

2 1

[PATCH 1/2] Revert "ASoC: meson: axg-tdm-interface: manage formatters in trigger"
by Neil Armstrong 22 Apr '22

22 Apr '22

This reverts commit bf5e4887eeddb48480568466536aa08ec7f179a5 because the following and required commit e138233e56e9829e65b6293887063a1a3ccb2d68 causes the following system crash when using audio: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:282 Reported-by: Dmitry Shmidt <dimitrysh(a)google.com> Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 0c31934a9630..e076ced30025 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -351,29 +351,13 @@ static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, return 0; } -static int axg_tdm_iface_trigger(struct snd_pcm_substream *substream, - int cmd, +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct axg_tdm_stream *ts = - snd_soc_dai_get_dma_data(dai, substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - axg_tdm_stream_start(ts); - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_STOP: - axg_tdm_stream_stop(ts); - break; - default: - return -EINVAL; - } + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); - return 0; + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); } static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) @@ -413,8 +397,8 @@ static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_fmt = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, .hw_free = axg_tdm_iface_hw_free, - .trigger = axg_tdm_iface_trigger, }; /* TDM Backend DAIs */ -- 2.25.1

3 6

[PATCH V2 1/2] ASoC: dt-bindings: max98396: add amplifier driver
by Ryan Lee 22 Apr '22

22 Apr '22

From: Ryan Lee <ryans.lee(a)analog.com> This patch adds dt-bindings information for Analog Devices MAX98396 and MAX98397 Smart Amplifier. Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- Changes from v1: Fixed yamllint/dtschema/dtc warnings and errors .../bindings/sound/adi,max98396.yaml | 79 +++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,max98396.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000000..ec4c10c2598a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98396.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98396 Speaker Amplifier Device Tree Bindings + +maintainers: + - Ryan Lee <ryans.lee(a)analog.com> + +description: + The MAX98396 is a mono Class-DG speaker amplifier with I/V sense. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + The MAX98397 is a variant of MAX98396 with wide input supply range. + +properties: + compatible: + enum: + - adi,max98396 + - adi,max98397 + reg: + maxItems: 1 + description: I2C address of the device. + + adi,vmon-slot-no: + description: slot number of the voltage sense monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current sense monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,spkfb-slot-no: + description: slot number of speaker DSP monitor + $ref: "/schemas/types.yaml#/definitions/uint32" + minimum: 0 + maximum: 15 + default: 0 + + adi,interleave-mode: + description: + For cases where a single combined channel for the I/V sense data + is not sufficient, the device can also be configured to share + a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame + interleaved on a single output channel. + type: boolean + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98396: amplifier@39 { + compatible = "adi,max98396"; + reg = <0x39>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; -- 2.25.1

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[PATCH -next V2] sound/oss/dmasound: fix 'dmasound_setup' defined but not used
by Miles Chen 22 Apr '22

22 Apr '22

We observed: 'dmasound_setup' defined but not used error with COMPILER=gcc ARCH=m68k DEFCONFIG=allmodconfig build. Fix it by adding __maybe_unused to dmasound_setup. Error(s): sound/oss/dmasound/dmasound_core.c:1431:12: error: 'dmasound_setup' defined but not used [-Werror=unused-function] Fixes: 9dd7c46346ca ("sound/oss/dmasound: fix build when drivers are mixed =y/=m") Signed-off-by: Miles Chen <miles.chen(a)mediatek.com> Cc: Takashi Iwai <tiwai(a)suse.com> --- Change sinve v1: add fixes tag use __maybe_unused --- sound/oss/dmasound/dmasound_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 9c48f3a9e3d1..164335d3c200 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1428,7 +1428,7 @@ void dmasound_deinit(void) unregister_sound_dsp(sq_unit); } -static int dmasound_setup(char *str) +static int __maybe_unused dmasound_setup(char *str) { int ints[6], size; -- 2.18.0

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