* sashiko.dev review (Re: [PATCH v4 07/17] dmaengine: sh: rz-dmac: Save the start LM descriptor)
From: Claudiu Beznea @ 2026-04-20 12:37 UTC (permalink / raw)
To: vkoul, Frank.Li, lgirdwood, broonie, perex, tiwai, biju.das.jz,
prabhakar.mahadev-lad.rj, p.zabel, geert+renesas,
fabrizio.castro.jz, long.luu.ur
Cc: dmaengine, linux-kernel, linux-sound, linux-renesas-soc,
Claudiu Beznea
In-Reply-To: <20260411114303.2814115-8-claudiu.beznea.uj@bp.renesas.com>
> commit 3d17698f275248947e3d2df42293a89e89cafaa6
> Author: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
> dmaengine: sh: rz-dmac: Save the start LM descriptor
> Save the start LM descriptor to avoid looping through the entire
> channel's LM descriptor list when computing the residue. This avoids
> unnecessary iterations.
> > diff --git a/drivers/dma/sh/rz-dmac.c b/drivers/dma/sh/rz-dmac.c
> > index 6bea7c8c70537..0f871c0a28bd9 100644
> > --- a/drivers/dma/sh/rz-dmac.c
> > +++ b/drivers/dma/sh/rz-dmac.c
> [ ... ]
> > @@ -377,6 +380,7 @@ static void rz_dmac_prepare_descs_for_slave_sg(struct
rz_dmac_chan *channel)
> > }
> >
> > lmdesc = channel->lmdesc.tail;
> > + d->start_lmdesc = lmdesc;
> >
> > for (i = 0, sg = sgl; i < sg_len; i++, sg = sg_next(sg)) {
> > if (d->direction == DMA_DEV_TO_MEM) {
> This isn't a bug introduced by this patch, but is there a potential buffer
> overflow in the link descriptor ring due to a lack of bounds checking?
That can happen but this is not the scope of this series. Series is already long
enough. I will address this in a separate series after the current one will be
merged.
> The driver allocates a fixed-size ring buffer of 64 link descriptors
> (DMAC_NR_LMDESC). However, rz_dmac_prepare_descs_for_slave_sg() iterates over
> the provided scatterlist and increments the lmdesc pointer without ever
> verifying if the number of scatterlist elements exceeds 64.
> If a client submits a sufficiently large scatterlist, will it silently
> overflow the ring buffer, overwriting the descriptors it just wrote, and
> cause the DMA controller to read a corrupted list of operations?
> [ ... ]
> > @@ -693,9 +697,10 @@ rz_dmac_get_next_lmdesc(struct rz_lmdesc *base, struct
rz_lmdesc *lmdesc)
> > return next;
> > }
> >
> > -static u32 rz_dmac_calculate_residue_bytes_in_vd(struct rz_dmac_chan
*channel, u32 crla)
> > +static u32 rz_dmac_calculate_residue_bytes_in_vd(struct rz_dmac_chan *channel,
> > + struct rz_dmac_desc *desc, u32
crla)
> > {
> > - struct rz_lmdesc *lmdesc = channel->lmdesc.head;
> > + struct rz_lmdesc *lmdesc = desc->start_lmdesc;
> > struct dma_chan *chan = &channel->vc.chan;
> > struct rz_dmac *dmac = to_rz_dmac(chan->device);
> > u32 residue = 0, i = 0;
> Can this lead to a NULL pointer dereference?
> In rz_dmac_prep_slave_sg(), a new descriptor is immediately added to
> channel->ld_queue. If a previous transfer completes before the client calls
> dma_async_issue_pending(), rz_dmac_irq_handler_thread() attempts to start the
> next transfer from ld_queue.
> However, vchan_next_desc() returns NULL since the transfer hasn't been
> issued, causing rz_dmac_xfer_desc() to return 0 without executing the hardware
> preparation function. The IRQ handler sees the 0 return value and erroneously
> moves the unprepared descriptor to channel->ld_active.
> If dmaengine_tx_status() is subsequently called, rz_dmac_chan_get_residue()
> fetches this active descriptor. Because it was never prepared, the newly added
> desc->start_lmdesc pointer remains NULL.
> The loop while (lmdesc->nxla != crla) then immediately dereferences NULL.
> Note that this issue was fixed later in the series by commit 'dmaengine: sh:
> rz-dmac: Use virt-dma APIs for channel descriptor processing'.
All the above could happen and the scope of patch "dmaengine: sh:
rz-dmac: Use virt-dma APIs for channel descriptor processing" was to fix all
these subtle issues. The patch here is only a preparation patch to keep things
simple in "dmaengine: sh: rz-dmac: Use virt-dma APIs for channel descriptor
processing"
> Additionally, this isn't a bug introduced by this patch, but does the
> mathematical logic in the residue calculation produce incorrect results?
> The loop while (lmdesc->nxla != crla) in
> rz_dmac_calculate_residue_bytes_in_vd() locates the link descriptor preceding
> the currently executing one. The subsequent loop while (lmdesc->chcfg &
> CHCFG_DEM) then iterates from this preceding descriptor, adding lmdesc->tb to
> the residue.
> If the hardware is executing the second descriptor of a transfer, it adds the
> total size of the first (already completed) descriptor, the total size of the
> second (currently executing) descriptor, and crtb (the remaining bytes of the
> second descriptor). Does this mathematically double-count bytes and include
> already transferred data?
> Conversely, if executing the first descriptor, the first loop wraps around the
> entire ring buffer, encounters the final descriptor of the previous transfer,
> and since its CHCFG_DEM bit is cleared, the second loop immediately
> terminates. Does this completely omit all subsequent descriptors from the
> residue calculation?
I think this is a valid point that I also noticed and thought about it. However,
I would like to keep it aside from this series, as it is already too big.
> [ ... ]
> This isn't a bug introduced by this patch, but is there a pre-existing race
> condition with dma_async_issue_pending()?
> rz_dmac_issue_pending() unconditionally calls rz_dmac_xfer_desc() if
> vchan_issue_pending() returns true, without checking if a transfer is already
> actively executing in hardware (e.g., if channel->desc is already set).
> For RZ_DMAC_DESC_MEMCPY transfers, the driver does not advance
> channel->lmdesc.tail, meaning the newly issued transfer will silently
> overwrite the exact link descriptor currently being executed by the active
> hardware. Does this lead to memory corruption and undefined hardware behavior?
> Note that this is fixed later in the series by commit 'dmaengine: sh: rz-dmac:
> Use virt-dma APIs for channel descriptor processing' which properly checks
> !channel->desc.
The role of the pointed descriptors was to fix all these subtle issues.
^ permalink raw reply
* sashiko.dev review (Re: [PATCH v4 04/17] dmaengine: sh: rz-dmac: Use rz_dmac_disable_hw())
From: Claudiu Beznea @ 2026-04-20 12:34 UTC (permalink / raw)
To: vkoul, Frank.Li, lgirdwood, broonie, perex, tiwai, biju.das.jz,
prabhakar.mahadev-lad.rj, p.zabel, geert+renesas,
fabrizio.castro.jz, long.luu.ur
Cc: dmaengine, linux-kernel, linux-sound, linux-renesas-soc,
Claudiu Beznea
In-Reply-To: <20260411114303.2814115-5-claudiu.beznea.uj@bp.renesas.com>
> commit f7b3ac8bceb85f09d10e2c6b3d49ae768415c77c
> Author: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
> dmaengine: sh: rz-dmac: Use rz_dmac_disable_hw()
> This patch unifies the hardware disable code by replacing an open-coded
> register write with rz_dmac_disable_hw(), moving it after vchan_init()
> so the channel device is properly initialized.
> > diff --git a/drivers/dma/sh/rz-dmac.c b/drivers/dma/sh/rz-dmac.c
> > index 1717b407ab9e9..40ddf534c0948 100644
> > --- a/drivers/dma/sh/rz-dmac.c
> > +++ b/drivers/dma/sh/rz-dmac.c
> [ ... ]
> > @@ -1000,15 +1000,15 @@ static int rz_dmac_chan_probe(struct rz_dmac *dmac,
> > }
> > rz_lmdesc_setup(channel, lmdesc);
> >
> > - /* Initialize register for each channel */
> > - rz_dmac_ch_writel(channel, CHCTRL_DEFAULT, CHCTRL, 1);
> > -
> > channel->vc.desc_free = rz_dmac_virt_desc_free;
> > vchan_init(&channel->vc, &dmac->engine);
> This is not a bug introduced by this patch, but seeing the vchan
> initialization here made me wonder about the driver teardown path.
> When vchan_init() is called, it sets up a background tasklet for the
> channel. During driver removal in rz_dmac_remove(), the DMA device is
> unregistered, but it does not appear that tasklet_kill() is ever called
> for this tasklet.
> Because the channel structures are allocated with devm_kzalloc, their
> memory is automatically freed immediately after rz_dmac_remove() returns.
> If a DMA transfer is active when the driver is forcefully unbound, could
> the IRQ thread schedule the tasklet via vchan_cookie_complete() right
> before the IRQ is freed by devres?
> Since free_irq() waits for the IRQ thread to finish but does not flush
> tasklets, could this scheduled tasklet execute after the device memory
> has been freed, resulting in a use-after-free?
> Should the driver explicitly kill the tasklet before returning from
> rz_dmac_remove()?
> > INIT_LIST_HEAD(&channel->ld_queue);
> > INIT_LIST_HEAD(&channel->ld_free);
> > INIT_LIST_HEAD(&channel->ld_active);
tasklet_setup() only sets the members of a struct tasklet object. The tasklet
itself is scheduled at runtime when the DMA clients requests transfers. On
remove the DMA clients would have already been called the
rz_dmac_terminate_all() and/or rz_dmac_device_synchronize() which should kill
the previously scheduled tasklet.
^ permalink raw reply
* sashiko.dev review (Re: [PATCH v4 01/17] dmaengine: sh: rz-dmac: Move interrupt request after everything is set up)
From: Claudiu Beznea @ 2026-04-20 12:33 UTC (permalink / raw)
To: vkoul, Frank.Li, lgirdwood, broonie, perex, tiwai, biju.das.jz,
prabhakar.mahadev-lad.rj, p.zabel, geert+renesas,
fabrizio.castro.jz, long.luu.ur
Cc: dmaengine, linux-kernel, linux-sound, linux-renesas-soc,
Claudiu Beznea, stable
In-Reply-To: <20260411114303.2814115-2-claudiu.beznea.uj@bp.renesas.com>
> commit b3dfdd5fbd2463db407b1396ee1ae79801cb96bf
> Author: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
> dmaengine: sh: rz-dmac: Move interrupt request after everything is set up
> This patch moves the channel interrupt request to the end of the channel
> probe function to prevent the IRQ handler from accessing uninitialized
> data. It also replaces dma_alloc_coherent() with dmam_alloc_coherent()
> to simplify the error path.
> > diff --git a/drivers/dma/sh/rz-dmac.c b/drivers/dma/sh/rz-dmac.c
> > index 625ff29024de0..9f206a33dcc6b 100644
> > --- a/drivers/dma/sh/rz-dmac.c
> > +++ b/drivers/dma/sh/rz-dmac.c
> [ ... ]
> > @@ -1210,16 +1207,6 @@ static int rz_dmac_probe(struct platform_device *pdev)
> > dma_register_err:
> > of_dma_controller_free(pdev->dev.of_node);
> > err:
> > - channel_num = i ? i - 1 : 0;
> > - for (i = 0; i < channel_num; i++) {
> > - struct rz_dmac_chan *channel = &dmac->channels[i];
> > -
> > - dma_free_coherent(&pdev->dev,
> > - sizeof(struct rz_lmdesc) * DMAC_NR_LMDESC,
> > - channel->lmdesc.base,
> > - channel->lmdesc.base_dma);
> > - }
> > -
> > reset_control_assert(dmac->rstc);
> > err_pm_runtime_put:
> > pm_runtime_put(&pdev->dev);
> > @@ -1232,18 +1219,9 @@ static int rz_dmac_probe(struct platform_device *pdev)
> > static void rz_dmac_remove(struct platform_device *pdev)
> > {
> > struct rz_dmac *dmac = platform_get_drvdata(pdev);
> > - unsigned int i;
> >
> > dma_async_device_unregister(&dmac->engine);
> > of_dma_controller_free(pdev->dev.of_node);
> > - for (i = 0; i < dmac->n_channels; i++) {
> > - struct rz_dmac_chan *channel = &dmac->channels[i];
> > -
> > - dma_free_coherent(&pdev->dev,
> > - sizeof(struct rz_lmdesc) * DMAC_NR_LMDESC,
> > - channel->lmdesc.base,
> > - channel->lmdesc.base_dma);
> > - }
> > reset_control_assert(dmac->rstc);
> > pm_runtime_put(&pdev->dev);
> > pm_runtime_disable(&pdev->dev);
> > }
> While this patch fixes the initialization races, does it leave a similar
> vulnerability exposed during teardown?
> Since devm_request_threaded_irq() is used to allocate the interrupts, they
> will remain active until after rz_dmac_remove() and the rz_dmac_probe() error
> paths complete.
> If an interrupt fires during or just after rz_dmac_remove(), could the handler
> attempt to access hardware registers while the device is in reset or powered
> down by pm_runtime_put()?
> Would it be safer to explicitly free or disable the IRQs before asserting the
> hardware reset and disabling runtime PM, or perhaps manage the reset and PM
> states via devm actions to guarantee correct teardown ordering?
On either failure or remove path the device is with runtime PM put (clocks being
disabled) and in reset state. The controller cannot generate interrupts from
this state.
However, to keep this series simple, I'll add this in a cleanup patch after the
current series will be merged.
^ permalink raw reply
* [PATCH v2 3/3] ASoC: codecs: lpass-wsa-macro: Use devm_clk_hw_register() for MCLK output
From: Ajay Kumar Nandam @ 2026-04-20 12:28 UTC (permalink / raw)
To: Srinivas Kandagatla, Liam Girdwood, Mark Brown, Jaroslav Kysela,
Takashi Iwai
Cc: linux-sound, linux-arm-msm, linux-kernel, mohammad.rafi.shaik,
ajay.nandam
In-Reply-To: <20260420122843.327171-1-ajay.nandam@oss.qualcomm.com>
The WSA macro driver registers the MCLK output clock using
clk_hw_register(), but does not explicitly unregister it in the remove
path or on probe failure.
Switch to devm_clk_hw_register() to make the registration resource-managed
so the clk_hw is automatically unregistered when the device is unbound or
probe fails. This avoids lifetime and cleanup issues and simplifies error
handling.
No functional change intended.
Signed-off-by: Ajay Kumar Nandam <ajay.nandam@oss.qualcomm.com>
---
sound/soc/codecs/lpass-wsa-macro.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index ded1cd8db831..29c8edfa739d 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -2657,7 +2657,7 @@ static int wsa_macro_register_mclk_output(struct wsa_macro *wsa)
init.num_parents = 1;
wsa->hw.init = &init;
hw = &wsa->hw;
- ret = clk_hw_register(wsa->dev, hw);
+ ret = devm_clk_hw_register(wsa->dev, hw);
if (ret)
return ret;
--
2.34.1
^ permalink raw reply related
* [PATCH v2 2/3] ASoC: codecs: lpass-va-macro: Switch to PM clock framework for runtime PM
From: Ajay Kumar Nandam @ 2026-04-20 12:28 UTC (permalink / raw)
To: Srinivas Kandagatla, Liam Girdwood, Mark Brown, Jaroslav Kysela,
Takashi Iwai
Cc: linux-sound, linux-arm-msm, linux-kernel, mohammad.rafi.shaik,
ajay.nandam
In-Reply-To: <20260420122843.327171-1-ajay.nandam@oss.qualcomm.com>
Convert the LPASS VA macro codec driver to use the PM clock framework
for runtime power management.
The driver now relies on pm_clk helpers and runtime PM instead of
manually enabling and disabling macro, dcodec, mclk, and npl clocks.
All clock control during runtime suspend and resume is delegated to
the PM core via pm_clk_suspend() and pm_clk_resume().
This change ensures clocks are only enabled when the VA macro is
active, improves power efficiency on LPASS platforms supporting
LPI/island modes, and aligns the driver with common ASoC runtime PM
patterns used across Qualcomm LPASS codec drivers.
Signed-off-by: Ajay Kumar Nandam <ajay.nandam@oss.qualcomm.com>
---
sound/soc/codecs/lpass-va-macro.c | 123 ++++++++++++++----------------
1 file changed, 58 insertions(+), 65 deletions(-)
diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c
index 528d5b167ecf..949275f1dfac 100644
--- a/sound/soc/codecs/lpass-va-macro.c
+++ b/sound/soc/codecs/lpass-va-macro.c
@@ -11,6 +11,7 @@
#include <linux/of_platform.h>
#include <linux/platform_device.h>
#include <linux/pm_runtime.h>
+#include <linux/pm_clock.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
#include <sound/soc.h>
@@ -1348,18 +1349,22 @@ static int fsgen_gate_enable(struct clk_hw *hw)
struct regmap *regmap = va->regmap;
int ret;
- if (va->has_swr_master) {
- ret = clk_prepare_enable(va->mclk);
- if (ret)
- return ret;
+ ret = pm_runtime_resume_and_get(va->dev);
+ if (ret < 0) {
+ pm_runtime_put_noidle(va->dev);
+ return ret;
}
ret = va_macro_mclk_enable(va, true);
+ if (ret) {
+ pm_runtime_put_noidle(va->dev);
+ return ret;
+ }
if (va->has_swr_master)
regmap_update_bits(regmap, CDC_VA_CLK_RST_CTRL_SWR_CONTROL,
CDC_VA_SWR_CLK_EN_MASK, CDC_VA_SWR_CLK_ENABLE);
- return ret;
+ return 0;
}
static void fsgen_gate_disable(struct clk_hw *hw)
@@ -1372,8 +1377,23 @@ static void fsgen_gate_disable(struct clk_hw *hw)
CDC_VA_SWR_CLK_EN_MASK, 0x0);
va_macro_mclk_enable(va, false);
- if (va->has_swr_master)
- clk_disable_unprepare(va->mclk);
+
+ pm_runtime_put_autosuspend(va->dev);
+}
+
+static int va_macro_setup_pm_clocks(struct device *dev, struct va_macro *va)
+{
+ int ret;
+
+ ret = devm_pm_clk_create(dev);
+ if (ret)
+ return ret;
+
+ ret = of_pm_clk_add_clks(dev);
+ if (ret < 0)
+ return ret;
+
+ return 0;
}
static int fsgen_gate_is_enabled(struct clk_hw *hw)
@@ -1534,6 +1554,7 @@ static int va_macro_probe(struct platform_device *pdev)
void __iomem *base;
u32 sample_rate = 0;
int ret;
+ int rpm_ret;
va = devm_kzalloc(dev, sizeof(*va), GFP_KERNEL);
if (!va)
@@ -1601,22 +1622,18 @@ static int va_macro_probe(struct platform_device *pdev)
clk_set_rate(va->npl, 2 * VA_MACRO_MCLK_FREQ);
}
- ret = clk_prepare_enable(va->macro);
- if (ret)
- goto err;
-
- ret = clk_prepare_enable(va->dcodec);
+ ret = va_macro_setup_pm_clocks(dev, va);
if (ret)
- goto err_dcodec;
+ goto err_rpm_disable;
- ret = clk_prepare_enable(va->mclk);
- if (ret)
- goto err_mclk;
+ pm_runtime_set_autosuspend_delay(dev, 100);
+ pm_runtime_use_autosuspend(dev);
+ pm_runtime_enable(dev);
- if (va->has_npl_clk) {
- ret = clk_prepare_enable(va->npl);
- if (ret)
- goto err_npl;
+ rpm_ret = pm_runtime_resume_and_get(dev);
+ if (rpm_ret < 0) {
+ ret = rpm_ret;
+ goto err_rpm_disable;
}
/**
@@ -1629,7 +1646,7 @@ static int va_macro_probe(struct platform_device *pdev)
/* read version from register */
ret = va_macro_set_lpass_codec_version(va);
if (ret)
- goto err_clkout;
+ goto err_rpm_put;
}
if (va->has_swr_master) {
@@ -1659,35 +1676,26 @@ static int va_macro_probe(struct platform_device *pdev)
va_macro_dais,
ARRAY_SIZE(va_macro_dais));
if (ret)
- goto err_clkout;
-
- pm_runtime_set_autosuspend_delay(dev, 3000);
- pm_runtime_use_autosuspend(dev);
- pm_runtime_mark_last_busy(dev);
- pm_runtime_set_active(dev);
- pm_runtime_enable(dev);
+ goto err_rpm_put;
ret = va_macro_register_fsgen_output(va);
if (ret)
- goto err_clkout;
+ goto err_rpm_put;
va->fsgen = devm_clk_hw_get_clk(dev, &va->hw, "fsgen");
if (IS_ERR(va->fsgen)) {
ret = PTR_ERR(va->fsgen);
- goto err_clkout;
+ goto err_rpm_put;
}
+ pm_runtime_put_autosuspend(dev);
+
return 0;
-err_clkout:
- if (va->has_npl_clk)
- clk_disable_unprepare(va->npl);
-err_npl:
- clk_disable_unprepare(va->mclk);
-err_mclk:
- clk_disable_unprepare(va->dcodec);
-err_dcodec:
- clk_disable_unprepare(va->macro);
+err_rpm_put:
+ pm_runtime_put_noidle(dev);
+err_rpm_disable:
+ pm_runtime_disable(dev);
err:
lpass_macro_pds_exit(va->pds);
@@ -1698,12 +1706,7 @@ static void va_macro_remove(struct platform_device *pdev)
{
struct va_macro *va = dev_get_drvdata(&pdev->dev);
- if (va->has_npl_clk)
- clk_disable_unprepare(va->npl);
-
- clk_disable_unprepare(va->mclk);
- clk_disable_unprepare(va->dcodec);
- clk_disable_unprepare(va->macro);
+ pm_runtime_disable(&pdev->dev);
lpass_macro_pds_exit(va->pds);
}
@@ -1715,12 +1718,7 @@ static int va_macro_runtime_suspend(struct device *dev)
regcache_cache_only(va->regmap, true);
regcache_mark_dirty(va->regmap);
- if (va->has_npl_clk)
- clk_disable_unprepare(va->npl);
-
- clk_disable_unprepare(va->mclk);
-
- return 0;
+ return pm_clk_suspend(dev);
}
static int va_macro_runtime_resume(struct device *dev)
@@ -1728,23 +1726,18 @@ static int va_macro_runtime_resume(struct device *dev)
struct va_macro *va = dev_get_drvdata(dev);
int ret;
- ret = clk_prepare_enable(va->mclk);
- if (ret) {
- dev_err(va->dev, "unable to prepare mclk\n");
+ ret = pm_clk_resume(dev);
+ if (ret)
return ret;
- }
-
- if (va->has_npl_clk) {
- ret = clk_prepare_enable(va->npl);
- if (ret) {
- clk_disable_unprepare(va->mclk);
- dev_err(va->dev, "unable to prepare npl\n");
- return ret;
- }
- }
regcache_cache_only(va->regmap, false);
- regcache_sync(va->regmap);
+
+ ret = regcache_sync(va->regmap);
+ if (ret) {
+ regcache_cache_only(va->regmap, true);
+ pm_clk_suspend(dev);
+ return ret;
+ }
return 0;
}
--
2.34.1
^ permalink raw reply related
* [PATCH v2 1/3] ASoC: codecs: lpass-wsa-macro: Switch to PM clock framework for runtime PM
From: Ajay Kumar Nandam @ 2026-04-20 12:28 UTC (permalink / raw)
To: Srinivas Kandagatla, Liam Girdwood, Mark Brown, Jaroslav Kysela,
Takashi Iwai
Cc: linux-sound, linux-arm-msm, linux-kernel, mohammad.rafi.shaik,
ajay.nandam
In-Reply-To: <20260420122843.327171-1-ajay.nandam@oss.qualcomm.com>
Convert the LPASS WSA macro codec driver to use the PM clock framework
for runtime power management.
The driver now relies on pm_clk helpers and runtime PM instead of
manually enabling and disabling macro, dcodec, mclk, npl, and fsgen
clocks. Runtime suspend and resume handling is delegated to the PM
core via pm_clk_suspend() and pm_clk_resume(), while existing runtime
PM callbacks continue to manage regcache state.
This ensures clocks are enabled only when the WSA macro is active,
improves power efficiency on LPASS platforms supporting LPI/island
modes, and aligns the driver with common ASoC runtime PM patterns used
across Qualcomm LPASS codec drivers.
Signed-off-by: Ajay Kumar Nandam <ajay.nandam@oss.qualcomm.com>
---
sound/soc/codecs/lpass-wsa-macro.c | 113 ++++++++++-------------------
1 file changed, 38 insertions(+), 75 deletions(-)
diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c
index b695c77c18ac..ded1cd8db831 100644
--- a/sound/soc/codecs/lpass-wsa-macro.c
+++ b/sound/soc/codecs/lpass-wsa-macro.c
@@ -14,6 +14,7 @@
#include <sound/soc-dapm.h>
#include <linux/pm_runtime.h>
#include <linux/of_platform.h>
+#include <linux/pm_clock.h>
#include <sound/tlv.h>
#include "lpass-macro-common.h"
@@ -2529,15 +2530,15 @@ static const struct snd_soc_dapm_route wsa_audio_map[] = {
static int wsa_swrm_clock(struct wsa_macro *wsa, bool enable)
{
struct regmap *regmap = wsa->regmap;
+ int ret;
- if (enable) {
- int ret;
+ ret = pm_runtime_get_sync(wsa->dev);
+ if (ret < 0) {
+ pm_runtime_put_noidle(wsa->dev);
+ return ret;
+ }
- ret = clk_prepare_enable(wsa->mclk);
- if (ret) {
- dev_err(wsa->dev, "failed to enable mclk\n");
- return ret;
- }
+ if (enable) {
wsa_macro_mclk_enable(wsa, true);
regmap_update_bits(regmap, CDC_WSA_CLK_RST_CTRL_SWR_CONTROL,
@@ -2548,9 +2549,9 @@ static int wsa_swrm_clock(struct wsa_macro *wsa, bool enable)
regmap_update_bits(regmap, CDC_WSA_CLK_RST_CTRL_SWR_CONTROL,
CDC_WSA_SWR_CLK_EN_MASK, 0);
wsa_macro_mclk_enable(wsa, false);
- clk_disable_unprepare(wsa->mclk);
}
+ pm_runtime_put_autosuspend(wsa->dev);
return 0;
}
@@ -2776,25 +2777,23 @@ static int wsa_macro_probe(struct platform_device *pdev)
clk_set_rate(wsa->mclk, WSA_MACRO_MCLK_FREQ);
clk_set_rate(wsa->npl, WSA_MACRO_MCLK_FREQ);
- ret = clk_prepare_enable(wsa->macro);
+ ret = devm_pm_clk_create(dev);
if (ret)
- goto err;
+ return ret;
- ret = clk_prepare_enable(wsa->dcodec);
- if (ret)
- goto err_dcodec;
+ ret = of_pm_clk_add_clks(dev);
+ if (ret < 0)
+ return ret;
- ret = clk_prepare_enable(wsa->mclk);
- if (ret)
- goto err_mclk;
+ pm_runtime_set_autosuspend_delay(dev, 100);
+ pm_runtime_use_autosuspend(dev);
+ pm_runtime_enable(dev);
- ret = clk_prepare_enable(wsa->npl);
- if (ret)
- goto err_npl;
- ret = clk_prepare_enable(wsa->fsgen);
- if (ret)
- goto err_fsgen;
+ ret = pm_runtime_resume_and_get(dev);
+ if (ret < 0) {
+ goto err_rpm_disable;
+ }
/* reset swr ip */
regmap_update_bits(wsa->regmap, CDC_WSA_CLK_RST_CTRL_SWR_CONTROL,
@@ -2811,44 +2810,25 @@ static int wsa_macro_probe(struct platform_device *pdev)
wsa_macro_dai,
ARRAY_SIZE(wsa_macro_dai));
if (ret)
- goto err_clkout;
-
- pm_runtime_set_autosuspend_delay(dev, 3000);
- pm_runtime_use_autosuspend(dev);
- pm_runtime_mark_last_busy(dev);
- pm_runtime_set_active(dev);
- pm_runtime_enable(dev);
+ goto err_rpm_put;
ret = wsa_macro_register_mclk_output(wsa);
if (ret)
- goto err_clkout;
+ goto err_rpm_put;
- return 0;
+ pm_runtime_put_autosuspend(dev);
-err_clkout:
- clk_disable_unprepare(wsa->fsgen);
-err_fsgen:
- clk_disable_unprepare(wsa->npl);
-err_npl:
- clk_disable_unprepare(wsa->mclk);
-err_mclk:
- clk_disable_unprepare(wsa->dcodec);
-err_dcodec:
- clk_disable_unprepare(wsa->macro);
-err:
+ return 0;
+err_rpm_put:
+ pm_runtime_put_noidle(dev);
+err_rpm_disable:
+ pm_runtime_disable(dev);
return ret;
-
}
static void wsa_macro_remove(struct platform_device *pdev)
{
- struct wsa_macro *wsa = dev_get_drvdata(&pdev->dev);
-
- clk_disable_unprepare(wsa->macro);
- clk_disable_unprepare(wsa->dcodec);
- clk_disable_unprepare(wsa->mclk);
- clk_disable_unprepare(wsa->npl);
- clk_disable_unprepare(wsa->fsgen);
+ pm_runtime_disable(&pdev->dev);
}
static int wsa_macro_runtime_suspend(struct device *dev)
@@ -2858,11 +2838,7 @@ static int wsa_macro_runtime_suspend(struct device *dev)
regcache_cache_only(wsa->regmap, true);
regcache_mark_dirty(wsa->regmap);
- clk_disable_unprepare(wsa->fsgen);
- clk_disable_unprepare(wsa->npl);
- clk_disable_unprepare(wsa->mclk);
-
- return 0;
+ return pm_clk_suspend(dev);
}
static int wsa_macro_runtime_resume(struct device *dev)
@@ -2870,34 +2846,21 @@ static int wsa_macro_runtime_resume(struct device *dev)
struct wsa_macro *wsa = dev_get_drvdata(dev);
int ret;
- ret = clk_prepare_enable(wsa->mclk);
+ ret = pm_clk_resume(dev);
if (ret) {
- dev_err(dev, "unable to prepare mclk\n");
+ regcache_cache_only(wsa->regmap, true);
return ret;
}
+ regcache_cache_only(wsa->regmap, false);
- ret = clk_prepare_enable(wsa->npl);
- if (ret) {
- dev_err(dev, "unable to prepare mclkx2\n");
- goto err_npl;
- }
-
- ret = clk_prepare_enable(wsa->fsgen);
+ ret = regcache_sync(wsa->regmap);
if (ret) {
- dev_err(dev, "unable to prepare fsgen\n");
- goto err_fsgen;
+ regcache_cache_only(wsa->regmap, true);
+ pm_clk_suspend(dev);
+ return ret;
}
- regcache_cache_only(wsa->regmap, false);
- regcache_sync(wsa->regmap);
-
return 0;
-err_fsgen:
- clk_disable_unprepare(wsa->npl);
-err_npl:
- clk_disable_unprepare(wsa->mclk);
-
- return ret;
}
static const struct dev_pm_ops wsa_macro_pm_ops = {
--
2.34.1
^ permalink raw reply related
* [PATCH v2 0/3] ASoC: qcom: lpass: Switch VA/WSA macros to PM clock framework
From: Ajay Kumar Nandam @ 2026-04-20 12:28 UTC (permalink / raw)
To: Srinivas Kandagatla, Liam Girdwood, Mark Brown, Jaroslav Kysela,
Takashi Iwai
Cc: linux-sound, linux-arm-msm, linux-kernel, mohammad.rafi.shaik,
ajay.nandam
This series converts LPASS WSA and VA macro codec drivers to the PM
clock framework for runtime PM clock handling.
Clock enable/disable sequencing during runtime suspend/resume is moved
to pm_clk helpers, while regcache state handling remains in the driver
runtime PM callbacks. This aligns these codec drivers with common LPASS
runtime PM patterns and reduces idle clock voting.
Patch 3 switches WSA MCLK clock registration to devm-managed
clk_hw registration.
---
v1: lore.kernel.org/r/20260413121824.375473-1-ajay.nandam@oss.qualcomm.com
Changes since v1:
- Fixed runtime resume error handling in WSA/VA paths:
- keep regcache cache-only on pm_clk_resume() failure
- unwind clocks with pm_clk_suspend() if regcache_sync() fails
- Dropped the separate "Guard optional NPL clock rate programming"
patch as discussed on list
Ajay Kumar Nandam (3):
ASoC: codecs: lpass-wsa-macro: Switch to PM clock framework for
runtime PM
ASoC: codecs: lpass-va-macro: Switch to PM clock framework for runtime
PM
ASoC: codecs: lpass-wsa-macro: Use devm_clk_hw_register() for MCLK
output
sound/soc/codecs/lpass-va-macro.c | 123 ++++++++++++++---------------
sound/soc/codecs/lpass-wsa-macro.c | 115 +++++++++------------------
2 files changed, 97 insertions(+), 141 deletions(-)
--
2.34.1
^ permalink raw reply
* [PATCH v2 RESEND] ASoC: sdw_utils: cs42l43: allow spk component names to be combined
From: Maciej Strozek @ 2026-04-20 11:48 UTC (permalink / raw)
To: Mark Brown, Jaroslav Kysela, Aaron Ma
Cc: Takashi Iwai, Bard Liao, linux-kernel, linux-sound, patches,
Maciej Strozek
Move handling of cs42l43-spk component string into SOF mechanism [1]
which will allow it to be aggregated with other speakers.
Likewise handle the cs35l56-bridge special case which should not be
combined to keep compatibility with UCM.
Link: https://github.com/thesofproject/linux/pull/5445 [1]
Link: https://github.com/alsa-project/alsa-ucm-conf/pull/747
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
---
Changes in v2: use dai->component instead of snd_soc_rtd_to_codec(rtd, 0)->component
---
sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 6 ------
sound/soc/sdw_utils/soc_sdw_cs42l43.c | 12 +-----------
sound/soc/sdw_utils/soc_sdw_utils.c | 20 ++++++++++++++++----
3 files changed, 17 insertions(+), 21 deletions(-)
diff --git a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
index 2a7109d53cbe3..e0e32a279787c 100644
--- a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
+++ b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
@@ -40,12 +40,6 @@ static int asoc_sdw_bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s spk:cs35l56-bridge",
- card->components);
- if (!card->components)
- return -ENOMEM;
-
ret = snd_soc_dapm_new_controls(dapm, bridge_widgets,
ARRAY_SIZE(bridge_widgets));
if (ret) {
diff --git a/sound/soc/sdw_utils/soc_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
index 4a451b9d4f137..e99ea3c4e5dde 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
@@ -107,21 +107,11 @@ EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_hs_rtd_init, "SND_SOC_SDW_UTILS");
int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
- struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
+ struct snd_soc_component *component = dai->component;
struct snd_soc_card *card = rtd->card;
struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
- struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
int ret;
- if (!(ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS)) {
- /* Will be set by the bridge code in this case */
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s spk:cs42l43-spk",
- card->components);
- if (!card->components)
- return -ENOMEM;
- }
-
ret = snd_soc_limit_volume(card, "cs42l43 Speaker Digital Volume",
CS42L43_SPK_VOLUME_0DB);
if (ret)
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index 2807f536eef0c..1637cc3f3d598 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -758,6 +758,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
{
.direction = {true, false},
.codec_name = "cs42l43-codec",
+ .component_name = "cs42l43-spk",
.dai_name = "cs42l43-dp6",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
@@ -1104,6 +1105,7 @@ static int asoc_sdw_find_codec_info_dai_index(const struct asoc_sdw_codec_info *
int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
+ struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
struct asoc_sdw_codec_info *codec_info;
struct snd_soc_dai *dai;
@@ -1179,16 +1181,26 @@ int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
/* Generate the spk component string for card->components string */
if (codec_info->dais[dai_index].dai_type == SOC_SDW_DAI_TYPE_AMP &&
codec_info->dais[dai_index].component_name) {
+ const char *component;
+
+ /*
+ * For the special case of cs42l43 with sidecar amps, use only
+ * "cs35l56-bridge" as the component name in card->components
+ */
+ if (ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS &&
+ !strcmp(codec_info->dais[dai_index].component_name, "cs42l43-spk"))
+ component = "cs35l56-bridge";
+ else
+ component = codec_info->dais[dai_index].component_name;
+
if (strlen (spk_components) == 0)
spk_components =
- devm_kasprintf(card->dev, GFP_KERNEL, "%s",
- codec_info->dais[dai_index].component_name);
+ devm_kasprintf(card->dev, GFP_KERNEL, "%s", component);
else
/* Append component name to spk_components */
spk_components =
devm_kasprintf(card->dev, GFP_KERNEL,
- "%s+%s", spk_components,
- codec_info->dais[dai_index].component_name);
+ "%s+%s", spk_components, component);
}
codec_info->dais[dai_index].rtd_init_done = true;
--
2.47.3
^ permalink raw reply related
* Re: [PATCH] ASoC: sdw_utils: cs42l43: allow spk component names to be combined
From: Maciej Strozek @ 2026-04-20 11:29 UTC (permalink / raw)
To: Mark Brown
Cc: Jaroslav Kysela, Aaron Ma, Takashi Iwai, Bard Liao, linux-kernel,
linux-sound, patches
In-Reply-To: <faf2f908-cff3-4efd-a80b-531a2060be68@sirena.org.uk>
W dniu pon, 20.04.2026 o godzinie 12∶20 +0100, użytkownik Mark Brown
napisał:
> On Mon, Apr 20, 2026 at 11:37:14AM +0100, Maciej Strozek wrote:
> > ---
> > Changes in v2: use dai->component instead of
> > snd_soc_rtd_to_codec(rtd, 0)->component
> > ---
>
> You say this is v2 but didn't actually send it as a v2 :(
Ahh sorry, missed the subject line, let me resend
--
Regards,
Maciej
^ permalink raw reply
* Re: [PATCH v9 1/4] ASoC: SDCA: Add PDE verification reusable helper
From: Pierre-Louis Bossart @ 2026-04-20 11:26 UTC (permalink / raw)
To: Charles Keepax
Cc: Niranjan H Y, linux-sound, linux-kernel, broonie, lgirdwood,
perex, tiwai, cezary.rojewski, peter.ujfalusi, yung-chuan.liao,
ranjani.sridharan, kai.vehmanen, baojun.xu, shenghao-ding,
sandeepk, v-hampiholi
In-Reply-To: <aeYBgE4JQiApVvRS@opensource.cirrus.com>
On 4/20/26 12:35, Charles Keepax wrote:
> On Mon, Apr 20, 2026 at 11:49:00AM +0200, Pierre-Louis Bossart wrote:
>> On 4/17/26 15:13, Niranjan H Y wrote:
>>> + * This function implements the polling logic but does NOT modify the power state.
>>> + * The caller is responsible for writing REQUESTED_PS before invoking this function.
>>
>> Erm, why not dealing with the write to REQUESTED_PS in this
>> helper? You have all the 'to' and 'from' information in the
>> parameters.
>
> I have no objections to moving that into the helper as well.
>
>>> + static const int polls = 100;
>>> + static const int default_poll_us = 1000;
>>> + unsigned int reg, val;
>>> + int i, poll_us = default_poll_us;
>>> + int ret;
>>> +
>>> + if (pde_delays && num_delays > 0) {
>>> + for (i = 0; i < num_delays; i++) {
>>> + if (pde_delays[i].from_ps == from_ps && pde_delays[i].to_ps == to_ps) {
>>> + poll_us = pde_delays[i].us / polls;
>>> + break;
>>> + }
>>> + }
>>> + }
>>> +
>>> + reg = SDW_SDCA_CTL(function_id, entity_id, SDCA_CTL_PDE_ACTUAL_PS, 0);
>>> +
>>> + for (i = 0; i < polls; i++) {
>>> + if (i)
>>> + fsleep(poll_us);
>>
>> This solution will loop for up to 100 times, and the sleep
>> duration could be questionable.
>
> The duration doesn't have to be precise here, as long as the
> result is longer than the requested time everything is fine.
>
>> Say for example you have a 10ms transition, do you really want
>> to read ACTUAL_PS every 100us?
>
> Quite potentially, I imagine it will be fairly common for parts
> to change PS a lot faster than the actual timeouts they provide,
> due to corner cases and people just being conservative in the
> DisCo. So its quite possible something that says 10mS typically
> switches in a couple 100uS.
>
>> If the pde_delay is 1ms then a read every 10us makes no sense,
>> the SoundWire command protocol would not be able to handle
>> such reads.
>>
>> A minimum threshold on poll_us would make sense IMHO.
>
> I guess you do reach a point where the soundwire command makes
> the delay effectively meaningless. What would you suggest for a
yep, that was the main point.
> minimum? Something like 100uS feels kinda reasonable to me,
> I would lean towards quite a small value here. Other options
> might be to look at some sort of exponential back off, doing the
> first few polls faster than later ones.
>
> This is definitely one of those situations where SDCA is a little
> too vague for its own good. But I would also say making a change
> like this should at a minimum be a separate patch rather than
> part of this one. And I am not convinced we need to block this
> series on updating it, although if we just wanted to go with a
> simple minimum that seems easy enough to add.
A minimum of 100us would be fine, we can always optimize for long delays later.
^ permalink raw reply
* Re: [PATCH] ASoC: sdw_utils: cs42l43: allow spk component names to be combined
From: Mark Brown @ 2026-04-20 11:20 UTC (permalink / raw)
To: Maciej Strozek
Cc: Jaroslav Kysela, Aaron Ma, Takashi Iwai, Bard Liao, linux-kernel,
linux-sound, patches
In-Reply-To: <20260420103722.4175356-1-mstrozek@opensource.cirrus.com>
[-- Attachment #1: Type: text/plain, Size: 222 bytes --]
On Mon, Apr 20, 2026 at 11:37:14AM +0100, Maciej Strozek wrote:
> ---
> Changes in v2: use dai->component instead of snd_soc_rtd_to_codec(rtd, 0)->component
> ---
You say this is v2 but didn't actually send it as a v2 :(
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply
* Re: [PATCH v3 0/2] ASoC: tegra210: simplify ADX/AMX byte map get/put logic
From: Mark Brown @ 2026-04-20 10:50 UTC (permalink / raw)
To: Piyush Patle
Cc: Liam Girdwood, Jaroslav Kysela, Takashi Iwai, Thierry Reding,
Jonathan Hunter, Sheetal, Kuninori Morimoto, linux-sound,
linux-tegra, linux-kernel
In-Reply-To: <CAMB+xkYZZprxmBMWnypYc3kfVa8ZN+WWrTOT9TpZqcQGOAyjVg@mail.gmail.com>
[-- Attachment #1: Type: text/plain, Size: 958 bytes --]
On Sat, Apr 18, 2026 at 07:06:36PM +0530, Piyush Patle wrote:
> On Sat, Apr 11, 2026 at 1:35 AM Piyush Patle <piyushpatle228@gmail.com> wrote:
> Just a gentle ping on this patch series.
Please don't send content free pings and please allow a reasonable time
for review. People get busy, go on holiday, attend conferences and so
on so unless there is some reason for urgency (like critical bug fixes)
please allow at least a couple of weeks for review. If there have been
review comments then people may be waiting for those to be addressed.
Sending content free pings adds to the mail volume (if they are seen at
all) which is often the problem and since they can't be reviewed
directly if something has gone wrong you'll have to resend the patches
anyway, so sending again is generally a better approach though there are
some other maintainers who like them - if in doubt look at how patches
for the subsystem are normally handled.
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply
* [PATCH] ASoC: sdw_utils: cs42l43: allow spk component names to be combined
From: Maciej Strozek @ 2026-04-20 10:37 UTC (permalink / raw)
To: Mark Brown, Jaroslav Kysela, Aaron Ma
Cc: Takashi Iwai, Bard Liao, linux-kernel, linux-sound, patches,
Maciej Strozek
Move handling of cs42l43-spk component string into SOF mechanism [1]
which will allow it to be aggregated with other speakers.
Likewise handle the cs35l56-bridge special case which should not be
combined to keep compatibility with UCM.
Link: https://github.com/thesofproject/linux/pull/5445 [1]
Link: https://github.com/alsa-project/alsa-ucm-conf/pull/747
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
---
Changes in v2: use dai->component instead of snd_soc_rtd_to_codec(rtd, 0)->component
---
sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 6 ------
sound/soc/sdw_utils/soc_sdw_cs42l43.c | 12 +-----------
sound/soc/sdw_utils/soc_sdw_utils.c | 20 ++++++++++++++++----
3 files changed, 17 insertions(+), 21 deletions(-)
diff --git a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
index 2a7109d53cbe3..e0e32a279787c 100644
--- a/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
+++ b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c
@@ -40,12 +40,6 @@ static int asoc_sdw_bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s spk:cs35l56-bridge",
- card->components);
- if (!card->components)
- return -ENOMEM;
-
ret = snd_soc_dapm_new_controls(dapm, bridge_widgets,
ARRAY_SIZE(bridge_widgets));
if (ret) {
diff --git a/sound/soc/sdw_utils/soc_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
index 4a451b9d4f137..e99ea3c4e5dde 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
@@ -107,21 +107,11 @@ EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_hs_rtd_init, "SND_SOC_SDW_UTILS");
int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
{
- struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
+ struct snd_soc_component *component = dai->component;
struct snd_soc_card *card = rtd->card;
struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
- struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
int ret;
- if (!(ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS)) {
- /* Will be set by the bridge code in this case */
- card->components = devm_kasprintf(card->dev, GFP_KERNEL,
- "%s spk:cs42l43-spk",
- card->components);
- if (!card->components)
- return -ENOMEM;
- }
-
ret = snd_soc_limit_volume(card, "cs42l43 Speaker Digital Volume",
CS42L43_SPK_VOLUME_0DB);
if (ret)
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index 2807f536eef0c..1637cc3f3d598 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -758,6 +758,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
{
.direction = {true, false},
.codec_name = "cs42l43-codec",
+ .component_name = "cs42l43-spk",
.dai_name = "cs42l43-dp6",
.dai_type = SOC_SDW_DAI_TYPE_AMP,
.dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
@@ -1104,6 +1105,7 @@ static int asoc_sdw_find_codec_info_dai_index(const struct asoc_sdw_codec_info *
int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
+ struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
struct asoc_sdw_codec_info *codec_info;
struct snd_soc_dai *dai;
@@ -1179,16 +1181,26 @@ int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
/* Generate the spk component string for card->components string */
if (codec_info->dais[dai_index].dai_type == SOC_SDW_DAI_TYPE_AMP &&
codec_info->dais[dai_index].component_name) {
+ const char *component;
+
+ /*
+ * For the special case of cs42l43 with sidecar amps, use only
+ * "cs35l56-bridge" as the component name in card->components
+ */
+ if (ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS &&
+ !strcmp(codec_info->dais[dai_index].component_name, "cs42l43-spk"))
+ component = "cs35l56-bridge";
+ else
+ component = codec_info->dais[dai_index].component_name;
+
if (strlen (spk_components) == 0)
spk_components =
- devm_kasprintf(card->dev, GFP_KERNEL, "%s",
- codec_info->dais[dai_index].component_name);
+ devm_kasprintf(card->dev, GFP_KERNEL, "%s", component);
else
/* Append component name to spk_components */
spk_components =
devm_kasprintf(card->dev, GFP_KERNEL,
- "%s+%s", spk_components,
- codec_info->dais[dai_index].component_name);
+ "%s+%s", spk_components, component);
}
codec_info->dais[dai_index].rtd_init_done = true;
--
2.47.3
^ permalink raw reply related
* Re: [PATCH v9 1/4] ASoC: SDCA: Add PDE verification reusable helper
From: Charles Keepax @ 2026-04-20 10:35 UTC (permalink / raw)
To: Pierre-Louis Bossart
Cc: Niranjan H Y, linux-sound, linux-kernel, broonie, lgirdwood,
perex, tiwai, cezary.rojewski, peter.ujfalusi, yung-chuan.liao,
ranjani.sridharan, kai.vehmanen, baojun.xu, shenghao-ding,
sandeepk, v-hampiholi
In-Reply-To: <6b72e996-a2dd-445b-b145-82644a6df8eb@linux.dev>
On Mon, Apr 20, 2026 at 11:49:00AM +0200, Pierre-Louis Bossart wrote:
> On 4/17/26 15:13, Niranjan H Y wrote:
> > + * This function implements the polling logic but does NOT modify the power state.
> > + * The caller is responsible for writing REQUESTED_PS before invoking this function.
>
> Erm, why not dealing with the write to REQUESTED_PS in this
> helper? You have all the 'to' and 'from' information in the
> parameters.
I have no objections to moving that into the helper as well.
> > + static const int polls = 100;
> > + static const int default_poll_us = 1000;
> > + unsigned int reg, val;
> > + int i, poll_us = default_poll_us;
> > + int ret;
> > +
> > + if (pde_delays && num_delays > 0) {
> > + for (i = 0; i < num_delays; i++) {
> > + if (pde_delays[i].from_ps == from_ps && pde_delays[i].to_ps == to_ps) {
> > + poll_us = pde_delays[i].us / polls;
> > + break;
> > + }
> > + }
> > + }
> > +
> > + reg = SDW_SDCA_CTL(function_id, entity_id, SDCA_CTL_PDE_ACTUAL_PS, 0);
> > +
> > + for (i = 0; i < polls; i++) {
> > + if (i)
> > + fsleep(poll_us);
>
> This solution will loop for up to 100 times, and the sleep
> duration could be questionable.
The duration doesn't have to be precise here, as long as the
result is longer than the requested time everything is fine.
> Say for example you have a 10ms transition, do you really want
> to read ACTUAL_PS every 100us?
Quite potentially, I imagine it will be fairly common for parts
to change PS a lot faster than the actual timeouts they provide,
due to corner cases and people just being conservative in the
DisCo. So its quite possible something that says 10mS typically
switches in a couple 100uS.
> If the pde_delay is 1ms then a read every 10us makes no sense,
> the SoundWire command protocol would not be able to handle
> such reads.
>
> A minimum threshold on poll_us would make sense IMHO.
I guess you do reach a point where the soundwire command makes
the delay effectively meaningless. What would you suggest for a
minimum? Something like 100uS feels kinda reasonable to me,
I would lean towards quite a small value here. Other options
might be to look at some sort of exponential back off, doing the
first few polls faster than later ones.
This is definitely one of those situations where SDCA is a little
too vague for its own good. But I would also say making a change
like this should at a minimum be a separate patch rather than
part of this one. And I am not convinced we need to block this
series on updating it, although if we just wanted to go with a
simple minimum that seems easy enough to add.
Thanks,
Charles
^ permalink raw reply
* Re: [PATCH v9 2/4] ASoC: tac5xx2-sdw: add soundwire based codec driver
From: Pierre-Louis Bossart @ 2026-04-20 10:10 UTC (permalink / raw)
To: Niranjan H Y, linux-sound
Cc: linux-kernel, broonie, ckeepax, lgirdwood, perex, tiwai,
cezary.rojewski, peter.ujfalusi, yung-chuan.liao,
ranjani.sridharan, kai.vehmanen, baojun.xu, shenghao-ding,
sandeepk, v-hampiholi
In-Reply-To: <20260417131401.3104-2-niranjan.hy@ti.com>
> +struct tac5xx2_prv {
> + struct snd_soc_component *component;
> + struct sdw_slave *sdw_peripheral;
> + struct sdca_function_data *sa_func_data;
> + struct sdca_function_data *sm_func_data;
> + struct sdca_function_data *uaj_func_data;
> + struct sdca_function_data *hid_func_data;
> + enum sdw_slave_status status;
> + /* Lock for firmware download and PDE state transitions.
> + * Serializes FW caching/download and DAPM-driven power
> + * state changes to prevent PDE operations during firmware load.
> + */
> + struct mutex pde_lock;
that's a lot of stuff that's protected with this lock. See below for one question...
> +static int tac_sdw_hw_params(struct snd_pcm_substream *substream,
> + struct snd_pcm_hw_params *params,
> + struct snd_soc_dai *dai)
> +{
> + struct snd_soc_component *component = dai->component;
> + struct tac5xx2_prv *tac_dev = snd_soc_component_get_drvdata(component);
> + struct sdw_stream_config stream_config = {0};
> + struct sdw_port_config port_config = {0};
> + struct sdw_stream_runtime *sdw_stream;
> + struct sdw_slave *sdw_peripheral = tac_dev->sdw_peripheral;
> + unsigned long time;
> + int ret;
> + int function_id;
> + int pde_entity;
> + int port_num;
> + u8 sample_rate_idx = 0;
> +
> + time = wait_for_completion_timeout(&sdw_peripheral->initialization_complete,
> + msecs_to_jiffies(TAC5XX2_PROBE_TIMEOUT_MS));
> + if (!time) {
> + dev_warn(tac_dev->dev, "%s: hw initialization timeout\n", __func__);
> + return -ETIMEDOUT;
> + }
> + if (!tac_dev->hw_init) {
> + dev_err(tac_dev->dev,
> + "error: operation without hw initialization");
> + return -EINVAL;
> + }
> +
> + sdw_stream = snd_soc_dai_get_dma_data(dai, substream);
> + if (!sdw_stream) {
> + dev_err(tac_dev->dev, "failed to get dma data");
> + return -EINVAL;
> + }
> +
> + ret = tac_clear_latch(tac_dev);
> + if (ret)
> + dev_warn(tac_dev->dev, "clear latch failed, err=%d", ret);
> +
> + switch (dai->id) {
> + case TAC5XX2_DMIC:
> + function_id = TAC_FUNCTION_ID_SM;
> + pde_entity = TAC_SDCA_ENT_PDE11;
> + port_num = TAC_SDW_PORT_NUM_DMIC;
> + break;
> + case TAC5XX2_UAJ:
> + function_id = TAC_FUNCTION_ID_UAJ;
> + pde_entity = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
> + TAC_SDCA_ENT_PDE47 : TAC_SDCA_ENT_PDE34;
> + port_num = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
> + TAC_SDW_PORT_NUM_UAJ_PLAYBACK :
> + TAC_SDW_PORT_NUM_UAJ_CAPTURE;
> + /* Detect and set jack type for UAJ path before playback.
> + * This is required as jack detection does not trigger interrupt
> + * when device is in runtime_pm suspend with bus in clock stop mode.
> + */
so here we have an interesting logic - or I misunderstood the comment?
If a headset is inserted when the device is in runtime_pm suspend, how would applications modify the routing and select playback on the headset, which would then ripple down to this hw_params() call?
IOW to play on a headset you first have to know there's a headset.
> + mutex_lock(&tac_dev->uaj_lock);
> + tac5xx2_sdca_headset_detect(tac_dev);
> + mutex_unlock(&tac_dev->uaj_lock);
> + break;
> + case TAC5XX2_SPK:
> + function_id = TAC_FUNCTION_ID_SA;
> + pde_entity = TAC_SDCA_ENT_PDE23;
> + port_num = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
> + TAC_SDW_PORT_NUM_SPK_PLAYBACK :
> + TAC_SDW_PORT_NUM_SPK_CAPTURE;
> + break;
> + default:
> + dev_err(tac_dev->dev, "Invalid dai id: %d", dai->id);
> + return -EINVAL;
> + }
> +
> + snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
> + port_config.num = port_num;
> + ret = sdw_stream_add_slave(sdw_peripheral, &stream_config,
> + &port_config, 1, sdw_stream);
> + if (ret) {
> + dev_err(dai->dev,
> + "Unable to configure port %d: %d\n", port_num, ret);
> + return ret;
> + }
> +
> + switch (params_rate(params)) {
> + case 48000:
> + sample_rate_idx = 0x01;
> + break;
> + case 44100:
> + sample_rate_idx = 0x02;
> + break;
> + case 96000:
> + sample_rate_idx = 0x03;
> + break;
> + case 88200:
> + sample_rate_idx = 0x04;
> + break;
> + default:
> + dev_dbg(tac_dev->dev, "Unsupported sample rate: %d Hz",
> + params_rate(params));
> + return -EINVAL;
> + }
> +
> + switch (function_id) {
> + case TAC_FUNCTION_ID_SM:
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(function_id, TAC_SDCA_ENT_CS113,
> + TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0),
> + sample_rate_idx);
> + if (ret) {
> + dev_err(tac_dev->dev, "Failed to set CS113 sample rate: %d", ret);
> + return ret;
> + }
> +
> + break;
> + case TAC_FUNCTION_ID_UAJ:
> + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(function_id, TAC_SDCA_ENT_CS41,
> + TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0),
> + sample_rate_idx);
> + if (ret) {
> + dev_err(tac_dev->dev, "Failed to set CS41 sample rate: %d", ret);
> + return ret;
> + }
> + } else {
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(function_id, TAC_SDCA_ENT_CS36,
> + TAC_SDCA_CTL_CS_SAMP_RATE_IDX, 0),
> + sample_rate_idx);
> + if (ret) {
> + dev_err(tac_dev->dev, "Failed to set CS36 sample rate: %d", ret);
> + return ret;
> + }
> + }
> + break;
> + case TAC_FUNCTION_ID_SA:
> + /* SmartAmp: no additional sample rate configuration needed */
> + break;
> + }
> +
> + guard(mutex)(&tac_dev->pde_lock);
question I mentioned above: when you reach the hw_params phase, do you really have a potential race with firmware download? What does this specific use of pde_lock protect against?
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(function_id, pde_entity,
> + TAC_SDCA_REQUESTED_PS, 0), 0);
> + if (ret) {
> + dev_err(tac_dev->dev, "failed to set PS to 0: %d\n", ret);
> + return ret;
> + }
> +
> + ret = sdca_asoc_pde_ensure_ps(tac_dev->dev, tac_dev->regmap,
> + function_id, pde_entity,
> + SDCA_PDE_PS3, SDCA_PDE_PS0,
> + NULL, 0);
> + if (ret)
> + dev_err(tac_dev->dev,
> + "failed to transition to PS0, err= %d\n", ret);
> + return ret;
> +}
> +
> +static s32 tac_sdw_pcm_hw_free(struct snd_pcm_substream *substream,
> + struct snd_soc_dai *dai)
> +{
> + s32 ret;
> + struct snd_soc_component *component = dai->component;
> + struct tac5xx2_prv *tac_dev =
> + snd_soc_component_get_drvdata(component);
> + struct sdw_stream_runtime *sdw_stream =
> + snd_soc_dai_get_dma_data(dai, substream);
> + int pde_entity, function_id;
> +
> + sdw_stream_remove_slave(tac_dev->sdw_peripheral, sdw_stream);
> +
> + switch (dai->id) {
> + case TAC5XX2_DMIC:
> + pde_entity = TAC_SDCA_ENT_PDE11;
> + function_id = TAC_FUNCTION_ID_SM;
> + break;
> + case TAC5XX2_UAJ:
> + function_id = TAC_FUNCTION_ID_UAJ;
> + pde_entity = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
> + TAC_SDCA_ENT_PDE47 : TAC_SDCA_ENT_PDE34;
> + break;
> + default:
> + function_id = TAC_FUNCTION_ID_SA;
> + pde_entity = TAC_SDCA_ENT_PDE23;
> + break;
> + }
> +
> + guard(mutex)(&tac_dev->pde_lock);
same here, do you really have a race with firmware download?
Or is this a case of dependencies between functions that requires all power state transitions to be serialized?
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(function_id, pde_entity, TAC_SDCA_REQUESTED_PS, 0),
> + SDCA_PDE_PS3);
> + if (ret)
> + return ret;
> +
> + ret = sdca_asoc_pde_ensure_ps(tac_dev->dev, tac_dev->regmap,
> + function_id, pde_entity,
> + SDCA_PDE_PS0, SDCA_PDE_PS3,
> + NULL, 0);
> + if (ret)
> + dev_err(tac_dev->dev, "failed to trasition from PS0 to PS3");
> + return ret;
> +}
> +
> +static const struct snd_soc_dai_ops tac_dai_ops = {
> + .hw_params = tac_sdw_hw_params,
> + .hw_free = tac_sdw_pcm_hw_free,
> + .set_stream = tac_set_sdw_stream,
> + .shutdown = tac_sdw_shutdown,
> +};
> +
> +static int tac5xx2_sdca_btn_type(unsigned char *buffer, struct tac5xx2_prv *tac_dev)
> +{
> + switch (*buffer) {
> + case 1: /* play pause */
> + return SND_JACK_BTN_0;
> + case 10: /* vol down */
> + return SND_JACK_BTN_3;
> + case 8: /* vol up */
> + return SND_JACK_BTN_2;
> + case 4: /* long press */
> + return SND_JACK_BTN_1;
> + case 2: /* next song */
> + case 32: /* next song */
> + return SND_JACK_BTN_4;
> + default:
> + return 0;
> + }
> +}
> +
> +static int tac5xx2_sdca_button_detect(struct tac5xx2_prv *tac_dev)
> +{
> + unsigned int btn_type, offset, idx;
> + int ret, value, owner;
> + u8 buf[2];
> +
> + ret = regmap_read(tac_dev->regmap,
> + SDW_SDCA_CTL(TAC_FUNCTION_ID_HID, TAC_SDCA_ENT_HID1,
> + TAC_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner);
> + if (ret) {
> + dev_err(tac_dev->dev,
> + "Failed to read current UMP message owner 0x%x", ret);
> + return ret;
> + }
> +
> + if (owner == SDCA_UMP_OWNER_DEVICE) {
> + dev_dbg(tac_dev->dev, "skip button detect as current owner is not host\n");
> + return 0;
> + }
> +
> + ret = regmap_read(tac_dev->regmap,
> + SDW_SDCA_CTL(TAC_FUNCTION_ID_HID, TAC_SDCA_ENT_HID1,
> + TAC_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset);
> + if (ret) {
> + dev_err(tac_dev->dev,
> + "Failed to read current UMP message offset: %d", ret);
> + goto end_btn_det;
> + }
> +
> + dev_dbg(tac_dev->dev, "button detect: message offset = %x", offset);
> +
> + for (idx = 0; idx < sizeof(buf); idx++) {
> + ret = regmap_read(tac_dev->regmap,
> + TAC_BUF_ADDR_HID1 + offset + idx, &value);
> + if (ret) {
> + dev_err(tac_dev->dev,
> + "Failed to read HID buffer: %d", ret);
> + goto end_btn_det;
> + }
> + buf[idx] = value & 0xff;
> + }
> +
> + if (buf[0] == 0x1) {
> + btn_type = tac5xx2_sdca_btn_type(&buf[1], tac_dev);
> + ret = btn_type;
> + }
> +
> +end_btn_det:
> + regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(TAC_FUNCTION_ID_HID, TAC_SDCA_ENT_HID1,
> + TAC_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01);
> +
> + return ret;
> +}
> +
> +static int tac5xx2_sdca_headset_detect(struct tac5xx2_prv *tac_dev)
> +{
> + int val, ret;
> +
> + if (!tac_has_uaj_support(tac_dev))
> + return 0;
can this really happen? usually you try to detect a headset if the device is capable of dealing with headsets, no?
Should this test be moved at a higher level before you enable low-level handling of headset stuff?
> + ret = regmap_read(tac_dev->regmap,
> + SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_GE35,
> + TAC_SDCA_CTL_DET_MODE, 0), &val);
> + if (ret) {
> + dev_err(tac_dev->dev, "Failed to read the detect mode");
> + return ret;
> + }
> +
> + switch (val) {
> + case 4:
> + tac_dev->jack_type = SND_JACK_MICROPHONE;
> + break;
> + case 5:
> + tac_dev->jack_type = SND_JACK_HEADPHONE;
> + break;
> + case 6:
> + tac_dev->jack_type = SND_JACK_HEADSET;
> + break;
> + case 0:
> + default:
> + tac_dev->jack_type = 0;
> + break;
> + }
> +
> + ret = regmap_write(tac_dev->regmap,
> + SDW_SDCA_CTL(TAC_FUNCTION_ID_UAJ, TAC_SDCA_ENT_GE35,
> + TAC_SDCA_CTL_SEL_MODE, 0), val);
> + if (ret)
> + dev_err(tac_dev->dev, "Failed to update the jack type to device");
> +
> + return 0;
> +}
> +
> +static int tac5xx2_set_jack(struct snd_soc_component *component,
> + struct snd_soc_jack *hs_jack, void *data)
> +{
> + struct tac5xx2_prv *tac_dev = snd_soc_component_get_drvdata(component);
> + int ret;
> +
> + if (!tac_has_uaj_support(tac_dev))
> + return 0;
same here, shouldn't set_jack() be added to the component callbacks before probe?
> + guard(mutex)(&tac_dev->uaj_lock);
> + if (!hs_jack) {
> + if (tac_dev->hs_jack) {
> + tac_dev->hs_jack = NULL;
> + ret = 0;
> + goto disable_interrupts;
> + }
> + return 0;
> + }
> +
> + tac_dev->hs_jack = hs_jack;
> + if (!tac_dev->hw_init) {
> + dev_err(tac_dev->dev, "jack init failed, hw not initialized");
> + return 0;
> + }
> +
> + ret = regmap_write(tac_dev->regmap, SDW_SCP_SDCA_INTMASK2,
> + SDW_SCP_SDCA_INTMASK_SDCA_11);
> + if (ret) {
> + dev_warn(tac_dev->dev,
> + "Failed to register jack detection interrupt");
> + goto disable_interrupts;
> + }
> +
> + ret = regmap_write(tac_dev->regmap, SDW_SCP_SDCA_INTMASK3,
> + SDW_SCP_SDCA_INTMASK_SDCA_16);
> + if (ret) {
> + dev_warn(tac_dev->dev,
> + "Failed to register for button detect interrupt");
> + goto disable_interrupts;
> + }
> +
> + return 0;
> +
> +disable_interrupts:
> + /* ignore errors while disabling interrupts */
> + regmap_write(tac_dev->regmap, SDW_SCP_SDCA_INTMASK2, 0);
> + regmap_write(tac_dev->regmap, SDW_SCP_SDCA_INTMASK3, 0);
> +
> + return ret;
> +}
> +static const struct snd_soc_component_driver soc_codec_driver_tacdevice = {
> + .probe = tac_component_probe,
> + .remove = tac_component_remove,
> + .controls = tac5xx2_snd_controls,
> + .num_controls = ARRAY_SIZE(tac5xx2_snd_controls),
> + .dapm_widgets = tac5xx2_common_widgets,
> + .num_dapm_widgets = ARRAY_SIZE(tac5xx2_common_widgets),
> + .dapm_routes = tac5xx2_common_routes,
> + .num_dapm_routes = ARRAY_SIZE(tac5xx2_common_routes),
> + .idle_bias_on = 0,
> + .endianness = 1,
> + .set_jack = tac5xx2_set_jack,
maybe make this dynamic and only populate .set_jack in tac_init() below when you can deal with a jack?
> +};
> +
> +static s32 tac_init(struct tac5xx2_prv *tac_dev)
> +{
> + s32 ret;
> + struct snd_soc_dai_driver *dai_drv;
> + int num_dais;
> +
> + dev_set_drvdata(tac_dev->dev, tac_dev);
> +
> + switch (tac_dev->part_id) {
> + case 0x5572:
> + dai_drv = tac5572_dai_driver;
> + num_dais = ARRAY_SIZE(tac5572_dai_driver);
> + break;
> + case 0x5672:
> + dai_drv = tac5672_dai_driver;
> + num_dais = ARRAY_SIZE(tac5672_dai_driver);
> + break;
> + case 0x5682:
> + dai_drv = tac5682_dai_driver;
> + num_dais = ARRAY_SIZE(tac5682_dai_driver);
> + break;
> + case 0x2883:
> + dai_drv = tas2883_dai_driver;
> + num_dais = ARRAY_SIZE(tas2883_dai_driver);
> + break;
> + default:
> + dev_err(tac_dev->dev, "Unsupported device: 0x%x\n",
> + tac_dev->part_id);
> + return -EINVAL;
> + }
> +
> + ret = devm_snd_soc_register_component(tac_dev->dev,
> + &soc_codec_driver_tacdevice,
> + dai_drv, num_dais);
> + if (ret) {
> + dev_err(tac_dev->dev, "%s: codec register error:%d.\n",
> + __func__, ret);
> + return ret;
> + }
> +
> + return 0;
> +}
^ permalink raw reply
* Re: [PATCH v9 1/4] ASoC: SDCA: Add PDE verification reusable helper
From: Pierre-Louis Bossart @ 2026-04-20 9:49 UTC (permalink / raw)
To: Niranjan H Y, linux-sound
Cc: linux-kernel, broonie, ckeepax, lgirdwood, perex, tiwai,
cezary.rojewski, peter.ujfalusi, yung-chuan.liao,
ranjani.sridharan, kai.vehmanen, baojun.xu, shenghao-ding,
sandeepk, v-hampiholi
In-Reply-To: <20260417131401.3104-1-niranjan.hy@ti.com>
On 4/17/26 15:13, Niranjan H Y wrote:
> Implement sdca_asoc_pde_ensure_ps() helper function to poll for PDE
> power state transitions. Per SDCA specification, after writing
> REQUESTED_PS, drivers must poll ACTUAL_PS until the target power state
> is reached.
Good initiative to introduce a new common helper...
> +/**
> + * sdca_asoc_pde_ensure_ps - Verify PDE power state reached target state
> + * @dev: Pointer to the device for error logging.
> + * @regmap: Register map for reading ACTUAL_PS register.
> + * @function_id: SDCA function identifier.
> + * @entity_id: SDCA entity identifier for the power domain.
> + * @from_ps: Source power state (SDCA_PDE_PSn value).
> + * @to_ps: Target power state (SDCA_PDE_PSn value).
> + * @pde_delays: Pointer to array of PDE delay specifications for this device,
> + * or NULL to use default polling interval.
> + * @num_delays: Number of entries in pde_delays array.
> + *
> + * This function polls the ACTUAL_PS register to verify that a PDE power state
> + * transition has completed. Per SDCA specification, after writing REQUESTED_PS,
> + * the caller must poll ACTUAL_PS until it reflects the requested state.
> + *
> + * This function implements the polling logic but does NOT modify the power state.
> + * The caller is responsible for writing REQUESTED_PS before invoking this function.
Erm, why not dealing with the write to REQUESTED_PS in this helper? You have all the 'to' and 'from' information in the parameters.
> + *
> + * If a delay table is provided, appropriate polling intervals are extracted based
> + * on the from_ps and to_ps transition. If no table is provided or no matching entry
> + * is found, a default polling interval is used.
> + *
> + * Return: Returns zero when ACTUAL_PS reaches the target state, -ETIMEDOUT if the
> + * polling times out before reaching the target state, or a negative error code if
> + * a register read fails.
> + */
> +int sdca_asoc_pde_ensure_ps(struct device *dev, struct regmap *regmap,
> + int function_id, int entity_id,
> + int from_ps, int to_ps,
> + const struct sdca_pde_delay *pde_delays,
> + int num_delays)
> +{
> + static const int polls = 100;
> + static const int default_poll_us = 1000;
> + unsigned int reg, val;
> + int i, poll_us = default_poll_us;
> + int ret;
> +
> + if (pde_delays && num_delays > 0) {
> + for (i = 0; i < num_delays; i++) {
> + if (pde_delays[i].from_ps == from_ps && pde_delays[i].to_ps == to_ps) {
> + poll_us = pde_delays[i].us / polls;
> + break;
> + }
> + }
> + }
> +
> + reg = SDW_SDCA_CTL(function_id, entity_id, SDCA_CTL_PDE_ACTUAL_PS, 0);
> +
> + for (i = 0; i < polls; i++) {
> + if (i)
> + fsleep(poll_us);
This solution will loop for up to 100 times, and the sleep duration could be questionable.
Say for example you have a 10ms transition, do you really want to read ACTUAL_PS every 100us?
If the pde_delay is 1ms then a read every 10us makes no sense, the SoundWire command protocol would not be able to handle such reads.
A minimum threshold on poll_us would make sense IMHO.
> +
> + ret = regmap_read(regmap, reg, &val);
> + if (ret)
> + return ret;
> + else if (val == to_ps)
> + return 0;
> + }
> +
> + dev_err(dev, "PDE power transition failed: expected 0x%x, got 0x%x\n", to_ps, val);
> + return -ETIMEDOUT;
> +}
> +EXPORT_SYMBOL(sdca_asoc_pde_ensure_ps);
^ permalink raw reply
* Re: [PATCH v9 1/4] ASoC: SDCA: Add PDE verification reusable helper
From: Charles Keepax @ 2026-04-20 9:57 UTC (permalink / raw)
To: Niranjan H Y
Cc: linux-sound, linux-kernel, broonie, lgirdwood, perex, tiwai,
cezary.rojewski, peter.ujfalusi, yung-chuan.liao,
ranjani.sridharan, kai.vehmanen, pierre-louis.bossart, baojun.xu,
shenghao-ding, sandeepk, v-hampiholi
In-Reply-To: <20260417131401.3104-1-niranjan.hy@ti.com>
On Fri, Apr 17, 2026 at 06:43:58PM +0530, Niranjan H Y wrote:
> Implement sdca_asoc_pde_ensure_ps() helper function to poll for PDE
> power state transitions. Per SDCA specification, after writing
> REQUESTED_PS, drivers must poll ACTUAL_PS until the target power state
> is reached.
>
> Changes include:
> - Add sdca_asoc_pde_ensure_ps() to handle ACTUAL_PS polling with
> support for device-specific delay tables or default intervals
> - Export function via sdca_asoc.h for use by SDCA-compliant drivers
> - Refactor entity_pde_event() in sdca_asoc.c to use the helper
>
> Signed-off-by: Niranjan H Y <niranjan.hy@ti.com>
> ---
> --- a/include/sound/sdca_asoc.h
> +++ b/include/sound/sdca_asoc.h
> @@ -99,4 +99,13 @@ int sdca_asoc_q78_put_volsw(struct snd_kcontrol *kcontrol,
> struct snd_ctl_elem_value *ucontrol);
> int sdca_asoc_q78_get_volsw(struct snd_kcontrol *kcontrol,
> struct snd_ctl_elem_value *ucontrol);
> +
> +struct device;
> +struct sdca_pde_delay;
sdca_pde_delay should go at the top of the file with the others,
device is already there so can be removed.
> +int sdca_asoc_pde_ensure_ps(struct device *dev, struct regmap *regmap,
Perhaps sdca_asoc_pde_poll_ps, but I don't feel super strongly on
that.
> + int function_id, int entity_id,
> + int from_ps, int to_ps,
> + const struct sdca_pde_delay *pde_delays,
> + int num_delays)
> +{
> + static const int polls = 100;
> + static const int default_poll_us = 1000;
> + unsigned int reg, val;
> + int i, poll_us = default_poll_us;
Put poll_us on its own line, and do all the assigned variables
first. I would combine i and ret on the same line instead.
Other than those minor nitpicks this looks good to me though.
Thanks,
Charles
^ permalink raw reply
* Re: [PATCH] ALSA: usb-audio: qcom: move offload usage counting to stream request handler
From: Guan-Yu Lin @ 2026-04-20 9:52 UTC (permalink / raw)
To: Takashi Iwai
Cc: gregkh, perex, kees, tiwai, quic_wcheng, arnd, wesley.cheng,
nkapron, linux-kernel, linux-sound
In-Reply-To: <87a4v2dla2.wl-tiwai@suse.de>
On Fri, Apr 17, 2026 at 4:49 PM Takashi Iwai <tiwai@suse.de> wrote:
>
> On Fri, 17 Apr 2026 07:43:30 +0200,
> Guan-Yu Lin wrote:
> >
> > Centralize usb_offload_get()/usb_offload_put() calls from various helper
> > functions to handle_uaudio_stream_req(), which ensures the usage count
> > is tied directly to the QMI stream enable/dsiable requests from audio
> > DSP. Such design provides a clear synchronization point for the
> > offloading lifecycle and correctly balances the count for both playback
> > and capture paths while avoiding redundant increments during auxiliary
> > operations.
> >
> > Suggested-by: Wesley Cheng <wesley.cheng@oss.qualcomm.com>
> > Signed-off-by: Guan-Yu Lin <guanyulin@google.com>
> > ---
> > sound/usb/qcom/qc_audio_offload.c | 20 ++++++++++----------
> > 1 file changed, 10 insertions(+), 10 deletions(-)
> >
> > diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c
> > index 2ac813d57f4f..cddab55d4691 100644
> > @@ -1605,6 +1597,12 @@ static void handle_uaudio_stream_req(struct qmi_handle *handle,
> > uadev[pcm_card_num].ctrl_intf = chip->ctrl_intf;
> >
> > if (req_msg->enable) {
> > + ret = usb_offload_get(subs->dev);
> > + if (ret < 0) {
> > + guard(mutex)(&chip->mutex);
> > + subs->opened = 0;
> > + goto response;
> > + }
> > ret = enable_audio_stream(subs,
> > map_pcm_format(req_msg->audio_format),
> > req_msg->number_of_ch, req_msg->bit_rate,
>
> I think the usb_offload_put() is missing at the error handling of the
> call above?
>
>
> thanks,
>
> Takashi
Yes, I think we should add error handling in this case. Let me update
it in the next version. Thanks.
Regards,
Guan-Yu
^ permalink raw reply
* [PATCH v9 4/5] ASoC: Intel: soc-acpi: arl: Add es9356 support
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
In-Reply-To: <20260420094349.14197-1-zhangyi@everest-semi.com>
Add support for the es9356 codec in the ARL board configuration.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
---
.../intel/common/soc-acpi-intel-arl-match.c | 52 +++++++++++++++++++
1 file changed, 52 insertions(+)
diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
index c952f7d2b..9782825dd 100644
--- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c
@@ -192,6 +192,42 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = {
},
};
+static const struct snd_soc_acpi_endpoint es9356_endpoints[] = {
+ { /* Jack Playback Endpoint */
+ .num = 0,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* DMIC Capture Endpoint */
+ .num = 1,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Jack Capture Endpoint */
+ .num = 2,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+ { /* Speaker Playback Endpoint */
+ .num = 3,
+ .aggregated = 0,
+ .group_position = 0,
+ .group_id = 0,
+ },
+};
+
+static const struct snd_soc_acpi_adr_device es9356_adr[] = {
+ {
+ .adr = 0x00013004b3935601ull,
+ .num_endpoints = ARRAY_SIZE(es9356_endpoints),
+ .endpoints = es9356_endpoints,
+ .name_prefix = "es9356"
+ }
+};
+
static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = {
{
.adr = 0x00003001FA424301ull,
@@ -255,6 +291,15 @@ static const struct snd_soc_acpi_adr_device rt1320_2_single_adr[] = {
}
};
+static const struct snd_soc_acpi_link_adr arl_n_mrd_es9356_link1[] = {
+ {
+ .mask = BIT(1),
+ .num_adr = ARRAY_SIZE(es9356_adr),
+ .adr_d = es9356_adr,
+ },
+ {}
+};
+
static const struct snd_soc_acpi_link_adr arl_cs42l43_l0[] = {
{
.mask = BIT(0),
@@ -528,6 +573,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_sdw_machines[] = {
.sof_tplg_filename = "sof-arl-rt722-l0_rt1320-l2.tplg",
.get_function_tplg_files = sof_sdw_get_tplg_files,
},
+ {
+ .link_mask = BIT(1),
+ .links = arl_n_mrd_es9356_link1,
+ .drv_name = "sof_sdw",
+ .sof_tplg_filename = "sof-arl-es9356.tplg",
+ .get_function_tplg_files = sof_sdw_get_tplg_files,
+ },
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_sdw_machines);
--
2.17.1
^ permalink raw reply related
* [PATCH v9 5/5] ASoC: Intel: sof_sdw: add es9356 support
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
In-Reply-To: <20260420094349.14197-1-zhangyi@everest-semi.com>
add Everest-semi ES9356 support
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
---
sound/soc/intel/boards/Kconfig | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index d53af8f7e..cddbd2aa4 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -532,6 +532,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
select MFD_CS42L43_SDW
select SND_SOC_CS35L56_SPI
select SND_SOC_CS35L56_SDW
+ select SND_SOC_ES9356
select SND_SOC_DMIC
select SND_SOC_INTEL_HDA_DSP_COMMON
imply SND_SOC_SDW_MOCKUP
--
2.17.1
^ permalink raw reply related
* [PATCH v9 1/5] ASoC: sdw_utils: add soc_sdw_es9356
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
In-Reply-To: <20260420094349.14197-1-zhangyi@everest-semi.com>
Add a utility program for handling ES9356 in the universal machine driver
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
---
include/sound/soc_sdw_utils.h | 14 ++
sound/soc/sdw_utils/Makefile | 1 +
sound/soc/sdw_utils/soc_sdw_es9356.c | 230 +++++++++++++++++++++++++++
3 files changed, 245 insertions(+)
create mode 100644 sound/soc/sdw_utils/soc_sdw_es9356.c
diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 489083183..5ddbbc8bb 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -223,6 +223,17 @@ int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card,
struct asoc_sdw_codec_info *info,
bool playback);
+/* es9356 codec support */
+int asoc_sdw_es9356_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct asoc_sdw_codec_info *info,
+ bool playback);
+int asoc_sdw_es9356_amp_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct asoc_sdw_codec_info *info,
+ bool playback);
+int asoc_sdw_es9356_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link);
+
/* CS AMP support */
int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card,
int *num_dais, int *num_devs);
@@ -274,5 +285,8 @@ int asoc_sdw_ti_amp_init(struct snd_soc_card *card,
int asoc_sdw_ti_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
int asoc_sdw_ti_amp_initial_settings(struct snd_soc_card *card,
const char *name_prefix);
+int asoc_sdw_es9356_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int asoc_sdw_es9356_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int asoc_sdw_es9356_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
#endif
diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile
index a8d091fd3..5ae8c69b8 100644
--- a/sound/soc/sdw_utils/Makefile
+++ b/sound/soc/sdw_utils/Makefile
@@ -8,6 +8,7 @@ snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \
soc_sdw_cs42l45.o \
soc_sdw_cs47l47.o \
soc_sdw_cs_amp.o \
+ soc_sdw_es9356.o \
soc_sdw_maxim.o \
soc_sdw_ti_amp.o
obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o
diff --git a/sound/soc/sdw_utils/soc_sdw_es9356.c b/sound/soc/sdw_utils/soc_sdw_es9356.c
new file mode 100644
index 000000000..aa405e7da
--- /dev/null
+++ b/sound/soc/sdw_utils/soc_sdw_es9356.c
@@ -0,0 +1,230 @@
+// SPDX-License-Identifier: GPL-2.0-only
+// Based on sof_sdw_rt5682.c
+// This file incorporates work covered by the following copyright notice:
+// Copyright (c) 2023 Intel Corporation
+// Copyright (c) 2024 Advanced Micro Devices, Inc.
+// Copyright (c) 2025 Everest Semiconductor Co., Ltd
+
+/*
+ * soc_sdw_es9356 - Helpers to handle ES9356 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/control.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <sound/soc_sdw_utils.h>
+
+/*
+ * Note this MUST be called before snd_soc_register_card(), so that the props
+ * are in place before the codec component driver's probe function parses them.
+ */
+static int es9356_add_codec_device_props(struct device *sdw_dev, unsigned long quirk)
+{
+ struct property_entry props[SOC_SDW_MAX_NO_PROPS] = {};
+ struct fwnode_handle *fwnode;
+ int ret;
+
+ if (!SOC_SDW_JACK_JDSRC(quirk))
+ return 0;
+ props[0] = PROPERTY_ENTRY_U32("everest,jd-src", SOC_SDW_JACK_JDSRC(quirk));
+
+ fwnode = fwnode_create_software_node(props, NULL);
+ if (IS_ERR(fwnode))
+ return PTR_ERR(fwnode);
+
+ ret = device_add_software_node(sdw_dev, to_software_node(fwnode));
+
+ fwnode_handle_put(fwnode);
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_route es9356_map[] = {
+ /* Headphones */
+ { "Headphone", NULL, "es9356 HP" },
+ { "es9356 MIC1", NULL, "Headset Mic" },
+};
+
+static const struct snd_soc_dapm_route es9356_spk_map[] = {
+ /* Speaker */
+ { "Speaker", NULL, "es9356 SPK" },
+};
+
+static const struct snd_soc_dapm_route es9356_dmic_map[] = {
+ /* DMIC */
+ { "es9356 PDM_DIN", NULL, "DMIC" },
+};
+
+static struct snd_soc_jack_pin es9356_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+int asoc_sdw_es9356_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s spk:es9356-spk",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_dapm_add_routes(dapm, es9356_spk_map,
+ ARRAY_SIZE(es9356_spk_map));
+ if (ret)
+ dev_err(card->dev, "es9356 map addition failed: %d\n", ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_spk_rtd_init, "SND_SOC_SDW_UTILS");
+
+int asoc_sdw_es9356_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
+ int ret;
+
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s mic:es9356-dmic",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_dapm_add_routes(dapm, es9356_dmic_map,
+ ARRAY_SIZE(es9356_dmic_map));
+ if (ret)
+ dev_err(card->dev, "es9356 map addition failed: %d\n", ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_dmic_rtd_init, "SND_SOC_SDW_UTILS");
+
+int asoc_sdw_es9356_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
+ struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct snd_soc_component *component;
+ struct snd_soc_jack *jack;
+ int ret;
+
+ component = dai->component;
+ card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+ "%s hs:es9356",
+ card->components);
+ if (!card->components)
+ return -ENOMEM;
+
+ ret = snd_soc_dapm_add_routes(dapm, es9356_map,
+ ARRAY_SIZE(es9356_map));
+
+ if (ret) {
+ dev_err(card->dev, "es9356 map addition failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+ SND_JACK_BTN_3 | SND_JACK_BTN_4,
+ &ctx->sdw_headset,
+ es9356_jack_pins,
+ ARRAY_SIZE(es9356_jack_pins));
+ if (ret) {
+ dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+ ret);
+ return ret;
+ }
+
+ jack = &ctx->sdw_headset;
+
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_NEXTSONG);
+ snd_jack_set_key(jack->jack, SND_JACK_BTN_4, KEY_PREVIOUSSONG);
+
+ ret = snd_soc_component_set_jack(component, jack, NULL);
+
+ if (ret)
+ dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+ ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_rtd_init, "SND_SOC_SDW_UTILS");
+
+int asoc_sdw_es9356_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link)
+{
+ struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
+
+ if (!ctx->headset_codec_dev)
+ return 0;
+
+ device_remove_software_node(ctx->headset_codec_dev);
+ put_device(ctx->headset_codec_dev);
+
+ return 0;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_exit, "SND_SOC_SDW_UTILS");
+
+int asoc_sdw_es9356_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct asoc_sdw_codec_info *info,
+ bool playback)
+{
+ struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card);
+ struct device *sdw_dev;
+ int ret;
+
+ /*
+ * headset should be initialized once.
+ * Do it with dai link for playback.
+ */
+ if (!playback)
+ return 0;
+
+ sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name);
+ if (!sdw_dev)
+ return -EPROBE_DEFER;
+
+ ret = es9356_add_codec_device_props(sdw_dev, ctx->mc_quirk);
+ if (ret < 0) {
+ put_device(sdw_dev);
+ return ret;
+ }
+ ctx->headset_codec_dev = sdw_dev;
+
+ return 0;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_init, "SND_SOC_SDW_UTILS");
+
+int asoc_sdw_es9356_amp_init(struct snd_soc_card *card,
+ struct snd_soc_dai_link *dai_links,
+ struct asoc_sdw_codec_info *info,
+ bool playback)
+{
+ if (!playback)
+ return 0;
+
+ info->amp_num++;
+
+ return 0;
+}
+EXPORT_SYMBOL_NS(asoc_sdw_es9356_amp_init, "SND_SOC_SDW_UTILS");
--
2.17.1
^ permalink raw reply related
* [PATCH v9 3/5] ASoC: es9356-sdca: Add ES9356 SDCA driver
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
In-Reply-To: <20260420094349.14197-1-zhangyi@everest-semi.com>
This is the codec driver for es9356-sdca.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
---
sound/soc/codecs/Kconfig | 7 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/es9356.c | 1154 +++++++++++++++++++++++++++++++++++++
sound/soc/codecs/es9356.h | 208 +++++++
4 files changed, 1371 insertions(+)
create mode 100644 sound/soc/codecs/es9356.c
create mode 100644 sound/soc/codecs/es9356.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index ca3e47db1..0cc6ffaec 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS
imply SND_SOC_ES8328_I2C
imply SND_SOC_ES8375
imply SND_SOC_ES8389
+ imply SND_SOC_ES9356
imply SND_SOC_ES7134
imply SND_SOC_ES7241
imply SND_SOC_FRAMER
@@ -1300,6 +1301,12 @@ config SND_SOC_ES8389
tristate "Everest Semi ES8389 CODEC"
depends on I2C
+config SND_SOC_ES9356
+ tristate "Everest Semi ES9356 CODEC SDW"
+ depends on SOUNDWIRE
+ select REGMAP_SOUNDWIRE
+ select REGMAP_SOUNDWIRE_MBQ
+
config SND_SOC_FRAMER
tristate "Framer codec"
depends on GENERIC_FRAMER
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 172861d17..91bdd8fa4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -138,6 +138,7 @@ snd-soc-es8328-i2c-y := es8328-i2c.o
snd-soc-es8328-spi-y := es8328-spi.o
snd-soc-es8375-y := es8375.o
snd-soc-es8389-y := es8389.o
+snd-soc-es9356-y := es9356.o
snd-soc-framer-y := framer-codec.o
snd-soc-fs-amp-lib-y := fs-amp-lib.o
snd-soc-fs210x-y := fs210x.o
@@ -575,6 +576,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
obj-$(CONFIG_SND_SOC_ES8375) += snd-soc-es8375.o
obj-$(CONFIG_SND_SOC_ES8389) += snd-soc-es8389.o
+obj-$(CONFIG_SND_SOC_ES9356) += snd-soc-es9356.o
obj-$(CONFIG_SND_SOC_FRAMER) += snd-soc-framer.o
obj-$(CONFIG_SND_SOC_FS_AMP_LIB)+= snd-soc-fs-amp-lib.o
obj-$(CONFIG_SND_SOC_FS210X) += snd-soc-fs210x.o
diff --git a/sound/soc/codecs/es9356.c b/sound/soc/codecs/es9356.c
new file mode 100644
index 000000000..b7834ce19
--- /dev/null
+++ b/sound/soc/codecs/es9356.c
@@ -0,0 +1,1154 @@
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// es9356.c -- SoundWire codec driver
+//
+// Copyright(c) 2025 Everest Semiconductor Co., Ltd
+//
+//
+
+#include <linux/device.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <linux/soundwire/sdw_registers.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <linux/pm_runtime.h>
+#include <sound/sdw.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/sdca_function.h>
+#include <sound/sdca_regmap.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+#include <sound/jack.h>
+#include <sound/sdca_asoc.h>
+#include "es9356.h"
+
+struct es9356_sdw_priv {
+ struct sdw_slave *slave;
+ struct device *dev;
+ struct regmap *regmap;
+ struct snd_soc_component *component;
+ struct snd_soc_jack *hs_jack;
+
+ struct mutex jack_lock;
+ struct mutex pde_lock;
+ bool hw_init;
+ bool first_hw_init;
+ int jack_type;
+ bool disable_irq;
+
+ struct delayed_work interrupt_handle_work;
+ struct delayed_work button_detect_work;
+ unsigned int sdca_status;
+};
+
+static int es9356_sdw_component_probe(struct snd_soc_component *component)
+{
+ struct es9356_sdw_priv *es9356 = snd_soc_component_get_drvdata(component);
+
+ es9356->component = component;
+
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -9600, 12, 0);
+static const DECLARE_TLV_DB_SCALE(amic_gain_tlv, 0, 3, 0);
+static const DECLARE_TLV_DB_SCALE(dmic_gain_tlv, 0, 6, 0);
+
+static const struct snd_kcontrol_new es9356_sdca_controls[] = {
+ SDCA_SINGLE_Q78_TLV("FU41 Left Playback Volume",
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_L),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU41 Right Playback Volume",
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_R),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU36 Left Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_L),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU36 Right Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_R),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU33 Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU33, ES9356_SDCA_CTL_FU_CH_GAIN, 0),
+ ES9356_GAIN_MIN, ES9356_AMIC_GAIN_MAX, ES9356_AMIC_GAIN_STEP, amic_gain_tlv),
+ SDCA_SINGLE_Q78_TLV("FU21 Left Playback Volume",
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_L),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU21 Right Playback Volume",
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_R),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU113 Left Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_L),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU113 Right Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_R),
+ ES9356_VOLUME_MIN, ES9356_VOLUME_MAX, ES9356_VOLUME_STEP, out_vol_tlv),
+ SDCA_SINGLE_Q78_TLV("FU11 Capture Volume",
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU11, ES9356_SDCA_CTL_FU_CH_GAIN, 0),
+ ES9356_GAIN_MIN, ES9356_DMIC_GAIN_MAX, ES9356_DMIC_GAIN_STEP, dmic_gain_tlv),
+};
+
+static const char *const es9356_left_mux_txt[] = {
+ "Left",
+ "Right",
+};
+
+static const char *const es9356_right_mux_txt[] = {
+ "Right",
+ "Left",
+};
+
+static const struct soc_enum es9356_left_mux_enum =
+ SOC_ENUM_SINGLE(ES9356_DAC_SWAP, 1,
+ ARRAY_SIZE(es9356_left_mux_txt), es9356_left_mux_txt);
+static const struct soc_enum es9356_right_mux_enum =
+ SOC_ENUM_SINGLE(ES9356_DAC_SWAP, 0,
+ ARRAY_SIZE(es9356_right_mux_txt), es9356_right_mux_txt);
+
+static const struct snd_kcontrol_new es9356_left_mux_controls =
+ SOC_DAPM_ENUM("Channel MUX", es9356_left_mux_enum);
+static const struct snd_kcontrol_new es9356_right_mux_controls =
+ SOC_DAPM_ENUM("Channel MUX", es9356_right_mux_enum);
+
+static const struct snd_soc_dapm_widget es9356_dapm_widgets[] = {
+ SND_SOC_DAPM_OUTPUT("HP"),
+ SND_SOC_DAPM_OUTPUT("SPK"),
+ SND_SOC_DAPM_INPUT("MIC1"),
+ SND_SOC_DAPM_INPUT("PDM_DIN"),
+
+ SND_SOC_DAPM_SUPPLY("DMIC Clock", ES9356_DMIC_GPIO, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_AIF_IN("DP4RX", "DP4 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("DP1TX", "DP1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+ SND_SOC_DAPM_PGA("IF DP3RXL", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("IF DP3RXR", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("Left Channel MUX", SND_SOC_NOPM, 0, 0, &es9356_left_mux_controls),
+ SND_SOC_DAPM_MUX("Right Channel MUX", SND_SOC_NOPM, 0, 0, &es9356_right_mux_controls),
+
+ SND_SOC_DAPM_DAC("FU 21 Left", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_MUTE, CH_L), 0, 1),
+ SND_SOC_DAPM_DAC("FU 21 Right", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_MUTE, CH_R), 0, 1),
+ SND_SOC_DAPM_DAC("FU 41 Left", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_MUTE, CH_L), 0, 1),
+ SND_SOC_DAPM_DAC("FU 41 Right", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_MUTE, CH_R), 0, 1),
+ SND_SOC_DAPM_DAC("FU 113 Left", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_MUTE, CH_L), 0, 1),
+ SND_SOC_DAPM_DAC("FU 113 Right", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_MUTE, CH_R), 0, 1),
+ SND_SOC_DAPM_DAC("FU 36 Left", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_MUTE, CH_L), 0, 1),
+ SND_SOC_DAPM_DAC("FU 36 Right", NULL,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_MUTE, CH_R), 0, 1),
+};
+
+static const struct snd_soc_dapm_route es9356_audio_map[] = {
+ {"FU 36 Left", NULL, "MIC1"},
+ {"FU 36 Right", NULL, "MIC1"},
+ {"DP2TX", NULL, "FU 36 Left"},
+ {"DP2TX", NULL, "FU 36 Right"},
+
+ {"PDM_DIN", NULL, "DMIC Clock"},
+ {"FU 113 Left", NULL, "PDM_DIN"},
+ {"FU 113 Right", NULL, "PDM_DIN"},
+ {"DP1TX", NULL, "FU 113 Left"},
+ {"DP1TX", NULL, "FU 113 Right"},
+
+ {"FU 41 Left", NULL, "DP4RX"},
+ {"FU 41 Right", NULL, "DP4RX"},
+
+ {"IF DP3RXL", NULL, "DP3RX"},
+ {"IF DP3RXR", NULL, "DP3RX"},
+
+ {"Left Channel MUX", "Left", "IF DP3RXL"},
+ {"Left Channel MUX", "Right", "IF DP3RXR"},
+ {"Right Channel MUX", "Left", "IF DP3RXL"},
+ {"Right Channel MUX", "Right", "IF DP3RXR"},
+
+ {"FU 21 Left", NULL, "Left Channel MUX"},
+ {"FU 21 Right", NULL, "Right Channel MUX"},
+
+ {"SPK", NULL, "FU 21 Left"},
+ {"SPK", NULL, "FU 21 Right"},
+
+ {"HP", NULL, "FU 41 Left"},
+ {"HP", NULL, "FU 41 Right"},
+};
+
+static int es9356_set_jack_detect(struct snd_soc_component *component,
+ struct snd_soc_jack *hs_jack, void *data)
+{
+ struct es9356_sdw_priv *es9356 = snd_soc_component_get_drvdata(component);
+ int ret;
+
+ es9356->hs_jack = hs_jack;
+ ret = pm_runtime_resume_and_get(component->dev);
+ if (ret < 0) {
+ if (ret != -EACCES) {
+ dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret);
+ return ret;
+ }
+ /* pm_runtime not enabled yet */
+ dev_info(component->dev, "%s: skipping jack init for now\n", __func__);
+ return 0;
+ }
+
+ if (es9356->hs_jack)
+ sdw_write_no_pm(es9356->slave, SDW_SCP_SDCA_INTMASK1,
+ (SDW_SCP_SDCA_INTMASK_SDCA_7 | SDW_SCP_SDCA_INTMASK_SDCA_5 | SDW_SCP_SDCA_INTMASK_SDCA_1));
+
+ pm_runtime_mark_last_busy(component->dev);
+ pm_runtime_put_autosuspend(component->dev);
+
+ return 0;
+}
+static const struct snd_soc_component_driver snd_soc_es9356_sdw_component = {
+ .probe = es9356_sdw_component_probe,
+ .controls = es9356_sdca_controls,
+ .num_controls = ARRAY_SIZE(es9356_sdca_controls),
+ .dapm_widgets = es9356_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(es9356_dapm_widgets),
+ .dapm_routes = es9356_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(es9356_audio_map),
+ .set_jack = es9356_set_jack_detect,
+ .endianness = 1,
+};
+
+static int es9356_sdw_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
+ int direction)
+{
+ snd_soc_dai_dma_data_set(dai, direction, sdw_stream);
+
+ return 0;
+}
+
+static void es9356_sdw_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct es9356_sdw_priv *es9356 = snd_soc_component_get_drvdata(component);
+
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static int es9356_sdca_button(unsigned int *buffer)
+{
+ int cur_button = -1;
+
+ if (*(buffer + 1) | *(buffer + 2))
+ return -EINVAL;
+ switch (*buffer) {
+ case 0x00:
+ cur_button = 0;
+ break;
+ case 0x20:
+ cur_button = SND_JACK_BTN_4;
+ break;
+ case 0x10:
+ cur_button = SND_JACK_BTN_2;
+ break;
+ case 0x08:
+ cur_button = SND_JACK_BTN_1;
+ break;
+ case 0x02:
+ cur_button = SND_JACK_BTN_3;
+ break;
+ case 0x01:
+ cur_button = SND_JACK_BTN_0;
+ break;
+ default:
+ break;
+ }
+
+ return cur_button;
+}
+
+static int es9356_sdca_button_detect(struct es9356_sdw_priv *es9356)
+{
+ unsigned int btn_type = 0, offset, idx, val, owner;
+ int ret;
+ unsigned int button[3];
+
+ ret = regmap_read(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_HID, ES9356_SDCA_ENT_HID01, ES9356_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner);
+ if (ret < 0 || owner == 0x01)
+ return 0;
+
+ ret = regmap_read(es9356->regmap, ES9356_BUF_ADDR_HID, &offset);
+ if (ret < 0)
+ goto button_det_end;
+
+ for (idx = 0; idx < ARRAY_SIZE(button); idx++) {
+ ret = regmap_read(es9356->regmap, ES9356_BUF_ADDR_HID + offset + idx, &val);
+ if (ret < 0)
+ goto button_det_end;
+ button[idx] = val;
+ }
+
+ btn_type = es9356_sdca_button(&button[0]);
+
+button_det_end:
+ if (owner == 0x00)
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_HID, ES9356_SDCA_ENT_HID01, ES9356_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01);
+
+ return btn_type;
+}
+
+static int es9356_sdca_headset_detect(struct es9356_sdw_priv *es9356)
+{
+ unsigned int reg;
+ int ret;
+
+ ret = regmap_read(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_GE35, ES9356_SDCA_CTL_DETECTED_MODE, 0), ®);
+
+ if (ret < 0)
+ goto io_error;
+
+ switch (reg) {
+ case 0x00:
+ es9356->jack_type = 0;
+ break;
+ case 0x03:
+ es9356->jack_type = SND_JACK_HEADPHONE;
+ break;
+ case 0x04:
+ es9356->jack_type = SND_JACK_HEADSET;
+ break;
+ default:
+ es9356->jack_type = 0;
+ return -1;
+ }
+
+ if (reg) {
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_GE35, ES9356_SDCA_CTL_SELECTED_MODE, 0), reg);
+ regmap_write(es9356->regmap, ES9356_HP_DETECTTIME, 0x75);
+ } else {
+ regmap_write(es9356->regmap, ES9356_HP_DETECTTIME, 0xa4);
+ }
+
+ return 0;
+
+io_error:
+ pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret);
+ return ret;
+}
+
+static void es9356_interrupt_handler(struct work_struct *work)
+{
+ struct es9356_sdw_priv *es9356 =
+ container_of(work, struct es9356_sdw_priv, interrupt_handle_work.work);
+ int ret, btn_type = 0;
+
+ if (!es9356->hs_jack)
+ return;
+
+ if (!es9356->component->card || !es9356->component->card->instantiated)
+ return;
+
+ /* Handling different types of interrupts based on the mask bit */
+ if (es9356->sdca_status & SDW_SCP_SDCA_INT_SDCA_7) {
+ btn_type = es9356_sdca_button_detect(es9356);
+ if (btn_type < 0)
+ return;
+ } else {
+ ret = es9356_sdca_headset_detect(es9356);
+ if (ret < 0)
+ return;
+ }
+
+ if (es9356->jack_type != SND_JACK_HEADSET)
+ btn_type = 0;
+
+ snd_soc_jack_report(es9356->hs_jack, es9356->jack_type | btn_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+ SND_JACK_BTN_4);
+
+ if (btn_type) {
+ snd_soc_jack_report(es9356->hs_jack, es9356->jack_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+ SND_JACK_BTN_4);
+ mod_delayed_work(system_power_efficient_wq,
+ &es9356->button_detect_work, msecs_to_jiffies(280));
+ }
+}
+
+static void es9356_button_detect_handler(struct work_struct *work)
+{
+ struct es9356_sdw_priv *es9356 =
+ container_of(work, struct es9356_sdw_priv, button_detect_work.work);
+ unsigned int reg, offset;
+ int ret, idx, btn_type = 0;
+ unsigned int button[3];
+
+ ret = regmap_read(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_GE35, ES9356_SDCA_CTL_DETECTED_MODE, 0), ®);
+
+ if (ret < 0)
+ goto io_error;
+
+ if (reg == 0x04) {
+ ret = regmap_read(es9356->regmap, ES9356_BUF_ADDR_HID, &offset);
+ if (ret < 0)
+ goto io_error;
+ for (idx = 0; idx < ARRAY_SIZE(button); idx++) {
+ ret = regmap_read(es9356->regmap, ES9356_BUF_ADDR_HID + offset + idx, ®);
+ if (ret < 0)
+ goto io_error;
+ button[idx] = reg;
+ }
+ btn_type = es9356_sdca_button(&button[0]);
+ if (btn_type < 0)
+ return;
+ }
+
+ snd_soc_jack_report(es9356->hs_jack, es9356->jack_type | btn_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+ SND_JACK_BTN_4);
+
+ if (btn_type) {
+ snd_soc_jack_report(es9356->hs_jack, es9356->jack_type,
+ SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+ SND_JACK_BTN_4);
+ mod_delayed_work(system_power_efficient_wq,
+ &es9356->button_detect_work, msecs_to_jiffies(280));
+ }
+
+ return;
+io_error:
+ pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret);
+}
+
+static int es9356_pde_transition_delay(struct es9356_sdw_priv *es9356, unsigned char func,
+ unsigned char entity, unsigned char ps)
+{
+ unsigned int retries = 10, val;
+
+ /* waiting for Actual PDE becomes to PS0/PS3 */
+ while (retries) {
+ regmap_read(es9356->regmap,
+ SDW_SDCA_CTL(func, entity, ES9356_SDCA_CTL_ACTUAL_POWER_STATE, 0), &val);
+ if (val == ps)
+ return 1;
+
+ usleep_range(1000, 1500);
+ retries--;
+ }
+ if (!retries) {
+ dev_dbg(&es9356->slave->dev, "%s PDE is NOT %s", __func__, ps?"PS3":"PS0");
+ }
+ return 0;
+}
+
+static int es9356_sdw_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct es9356_sdw_priv *es9356 = snd_soc_component_get_drvdata(component);
+ struct sdw_stream_config stream_config = {0};
+ struct sdw_port_config port_config = {0};
+ struct sdw_stream_runtime *sdw_stream = snd_soc_dai_get_dma_data(dai, substream);
+ int ret;
+ unsigned char func, pde_entity, cs_entity;
+ unsigned char ps0 = 0x0, ps3 = 0x3;
+ unsigned int rate;
+
+ if (!sdw_stream)
+ return -EINVAL;
+
+ if (!es9356->slave)
+ return -EINVAL;
+
+ /* SoundWire specific configuration */
+ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
+
+ port_config.num = dai->id;
+
+ ret = sdw_stream_add_slave(es9356->slave, &stream_config,
+ &port_config, 1, sdw_stream);
+ if (ret) {
+ dev_err(dai->dev, "Unable to configure port\n");
+ return -EINVAL;
+ }
+
+ switch (params_rate(params)) {
+ case 16000:
+ rate = ES9356_SDCA_RATE_16000HZ;
+ break;
+ case 44100:
+ rate = ES9356_SDCA_RATE_44100HZ;
+ break;
+ case 48000:
+ rate = ES9356_SDCA_RATE_48000HZ;
+ break;
+ case 96000:
+ rate = ES9356_SDCA_RATE_96000HZ;
+ break;
+ default:
+ dev_err(component->dev, "%s: Rate %d is not supported\n",
+ __func__, params_rate(params));
+ return -EINVAL;
+ }
+
+ switch (dai->id) {
+ case ES9356_DMIC:
+ func = FUNC_NUM_MIC;
+ cs_entity = ES9356_SDCA_ENT_CS113;
+ pde_entity = ES9356_SDCA_ENT_PDE11;
+ break;
+ case ES9356_AMP:
+ func = FUNC_NUM_AMP;
+ cs_entity = ES9356_SDCA_ENT_CS21;
+ pde_entity = ES9356_SDCA_ENT_PDE23;
+ break;
+ case ES9356_JACK_IN:
+ func = FUNC_NUM_UAJ;
+ cs_entity = ES9356_SDCA_ENT_CS36;
+ pde_entity = ES9356_SDCA_ENT_PDE34;
+ break;
+ case ES9356_JACK_OUT:
+ func = FUNC_NUM_UAJ;
+ cs_entity = ES9356_SDCA_ENT_CS41;
+ pde_entity = ES9356_SDCA_ENT_PDE47;
+ break;
+ default:
+ dev_err(component->dev, "%s: Invalid dai id: %d\n",
+ __func__, dai->id);
+ return -EINVAL;
+ }
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(func, cs_entity, ES9356_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), rate);
+
+ mutex_lock(&es9356->pde_lock);
+ ret = es9356_pde_transition_delay(es9356, func, pde_entity, ps3);
+ if (ret) {
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(func, pde_entity, ES9356_SDCA_CTL_REQ_POWER_STATE, 0), ps0);
+ ret = es9356_pde_transition_delay(es9356, func, pde_entity, ps0);
+ } else
+ dev_dbg(component->dev, "%s PDE is already PS0", __func__);
+
+ mutex_unlock(&es9356->pde_lock);
+
+ return 0;
+}
+
+static int es9356_sdw_pcm_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct es9356_sdw_priv *es9356 = snd_soc_component_get_drvdata(component);
+ struct sdw_stream_runtime *sdw_stream = snd_soc_dai_get_dma_data(dai, substream);
+ int ret;
+ unsigned char func, pde_entity;
+ unsigned char ps0 = 0x0, ps3 = 0x3;
+
+ if (!es9356->slave)
+ return -EINVAL;
+
+ sdw_stream_remove_slave(es9356->slave, sdw_stream);
+
+ switch (dai->id) {
+ case ES9356_DMIC:
+ func = FUNC_NUM_MIC;
+ pde_entity = ES9356_SDCA_ENT_PDE11;
+ break;
+ case ES9356_AMP:
+ func = FUNC_NUM_AMP;
+ pde_entity = ES9356_SDCA_ENT_PDE23;
+ break;
+ case ES9356_JACK_IN:
+ func = FUNC_NUM_UAJ;
+ pde_entity = ES9356_SDCA_ENT_PDE34;
+ break;
+ case ES9356_JACK_OUT:
+ func = FUNC_NUM_UAJ;
+ pde_entity = ES9356_SDCA_ENT_PDE47;
+ break;
+ default:
+ dev_err(component->dev, "%s: Invalid dai id: %d\n",
+ __func__, dai->id);
+ return -EINVAL;
+ }
+
+ mutex_lock(&es9356->pde_lock);
+ ret = es9356_pde_transition_delay(es9356, func, pde_entity, ps0);
+ if (ret) {
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(func, pde_entity, ES9356_SDCA_CTL_REQ_POWER_STATE, 0), ps3);
+ ret = es9356_pde_transition_delay(es9356, func, pde_entity, ps3);
+ } else
+ dev_dbg(component->dev, "%s PDE is already PS0", __func__);
+
+ mutex_unlock(&es9356->pde_lock);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops es9356_sdw_ops = {
+ .hw_params = es9356_sdw_pcm_hw_params,
+ .hw_free = es9356_sdw_pcm_hw_free,
+ .set_stream = es9356_sdw_set_sdw_stream,
+ .shutdown = es9356_sdw_shutdown,
+};
+
+static struct snd_soc_dai_driver es9356_sdw_dai[] = {
+ {
+ .name = "es9356-sdp-aif4",
+ .id = ES9356_DMIC,
+ .capture = {
+ .stream_name = "DP1 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &es9356_sdw_ops,
+ },
+ {
+ .name = "es9356-sdp-aif2",
+ .id = ES9356_JACK_IN,
+ .capture = {
+ .stream_name = "DP2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &es9356_sdw_ops,
+ },
+ {
+ .name = "es9356-sdp-aif3",
+ .id = ES9356_AMP,
+ .playback = {
+ .stream_name = "DP3 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &es9356_sdw_ops,
+ },
+ {
+ .name = "es9356-sdp-aif1",
+ .id = ES9356_JACK_OUT,
+ .playback = {
+ .stream_name = "DP4 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &es9356_sdw_ops,
+ },
+};
+
+static int es9356_sdca_mbq_size(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_L):
+ case SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_R):
+ case SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_L):
+ case SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_R):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_L):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_R):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_L):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_R):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU33, ES9356_SDCA_CTL_FU_CH_GAIN, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU11, ES9356_SDCA_CTL_FU_CH_GAIN, 0):
+ return 2;
+ default:
+ return 1;
+ }
+}
+
+static struct regmap_sdw_mbq_cfg es9356_mbq_config = {
+ .mbq_size = es9356_sdca_mbq_size,
+};
+
+static bool es9356_sdca_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case ES9356_BUF_ADDR_HID:
+ case ES9356_HID_BYTE2:
+ case ES9356_HID_BYTE3:
+ case ES9356_HID_BYTE4:
+ case SDW_SDCA_CTL(FUNC_NUM_HID, ES9356_SDCA_ENT_HID01, ES9356_SDCA_CTL_HIDTX_CURRENT_OWNER, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_GE35, ES9356_SDCA_CTL_DETECTED_MODE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_GE35, ES9356_SDCA_CTL_SELECTED_MODE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_PDE23, ES9356_SDCA_CTL_REQ_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_PDE23, ES9356_SDCA_CTL_ACTUAL_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_PDE11, ES9356_SDCA_CTL_REQ_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_PDE11, ES9356_SDCA_CTL_ACTUAL_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_PDE47, ES9356_SDCA_CTL_REQ_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_PDE47, ES9356_SDCA_CTL_ACTUAL_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_PDE34, ES9356_SDCA_CTL_REQ_POWER_STATE, 0):
+ case SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_PDE34, ES9356_SDCA_CTL_ACTUAL_POWER_STATE, 0):
+ case ES9356_FLAGS_HP:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config es9356_sdca_regmap = {
+ .reg_bits = 32,
+ .val_bits = 16,
+ .volatile_reg = es9356_sdca_volatile_register,
+ .max_register = 0x45ffffff,
+ .cache_type = REGCACHE_MAPLE,
+ .use_single_read = true,
+ .use_single_write = true,
+};
+
+static void es9356_register_init(struct es9356_sdw_priv *es9356)
+{
+ regmap_write(es9356->regmap, ES9356_STATE, 0x02);
+ regmap_write(es9356->regmap, ES9356_ENDPOINT_MODE, 0x24);
+ regmap_write(es9356->regmap, ES9356_PRE_DIV_CTL, 0x00);
+ regmap_write(es9356->regmap, ES9356_ADC_OSR, 0x18);
+ regmap_write(es9356->regmap, ES9356_ADC_OSRGAIN, 0x13);
+ regmap_write(es9356->regmap, ES9356_DAC_OSR, 0x16);
+ regmap_write(es9356->regmap, ES9356_CLK_CTL, 0x0f);
+ regmap_write(es9356->regmap, ES9356_CSM_RESET, 0x01);
+ regmap_write(es9356->regmap, ES9356_CLK_SEL, 0x30);
+
+ regmap_write(es9356->regmap, ES9356_DETCLK_CTL, 0x51);
+ regmap_write(es9356->regmap, ES9356_HP_TYPE, 0x10);
+ regmap_write(es9356->regmap, ES9356_MICBIAS_CTL, 0x10);
+ regmap_write(es9356->regmap, ES9356_HPDETECT_CTL, 0x07);
+ regmap_write(es9356->regmap, ES9356_ADC_ANA, 0x30);
+ regmap_write(es9356->regmap, ES9356_PGA_CTL, 0xa8);
+ regmap_write(es9356->regmap, ES9356_ADC_INT, 0xaa);
+ regmap_write(es9356->regmap, ES9356_ADC_LP, 0x19);
+ regmap_write(es9356->regmap, ES9356_VMID1SEL, 0xbc);
+ regmap_write(es9356->regmap, ES9356_VMID_TIME, 0x0b);
+ regmap_write(es9356->regmap, ES9356_STATE_TIME, 0xbb);
+ regmap_write(es9356->regmap, ES9356_HP_SPK_TIME, 0x77);
+ regmap_write(es9356->regmap, ES9356_HP_DETECTTIME, 0xa4);
+ regmap_write(es9356->regmap, ES9356_MICBIAS_SEL, 0x15);
+ regmap_write(es9356->regmap, ES9356_KEY_PRESS_TIME, 0xff);
+ regmap_write(es9356->regmap, ES9356_KEY_RELEASE_TIME, 0xff);
+ regmap_write(es9356->regmap, ES9356_KEY_HOLD_TIME, 0x0f);
+ regmap_write(es9356->regmap, ES9356_BTSEL_REF, 0x00);
+ regmap_write(es9356->regmap, ES9356_KEYD_DETECT, 0x18);
+ regmap_write(es9356->regmap, ES9356_MICBIAS_RES, 0x03);
+ regmap_write(es9356->regmap, ES9356_BUTTON_CHARGE, 0x00);
+ regmap_write(es9356->regmap, ES9356_CALIBRATION_TIME, 0x13);
+ regmap_write(es9356->regmap, ES9356_CALIBRATION_SETTING, 0xf4);
+
+ regmap_write(es9356->regmap, ES9356_SPK_VOLUME, 0x33);
+ regmap_write(es9356->regmap, ES9356_DAC_VROI, 0x01);
+ regmap_write(es9356->regmap, ES9356_DAC_LP, 0x00);
+ regmap_write(es9356->regmap, ES9356_HP_IBIAS, 0x04);
+ regmap_write(es9356->regmap, ES9356_HP_LP, 0x03);
+ regmap_write(es9356->regmap, ES9356_SPKLDO_CTL, 0x65);
+ regmap_write(es9356->regmap, ES9356_SPKBIAS_COMP, 0x09);
+ regmap_write(es9356->regmap, ES9356_VMID1STL, 0x00);
+ regmap_write(es9356->regmap, ES9356_VMID2STL, 0x00);
+ regmap_write(es9356->regmap, ES9356_VSEL, 0xfc);
+
+ regmap_write(es9356->regmap, ES9356_IBIASGEN, 0x10);
+ regmap_write(es9356->regmap, ES9356_ADC_AMIC_CTL, 0x0d);
+ regmap_write(es9356->regmap, ES9356_STATE, 0x0e);
+ regmap_write(es9356->regmap, ES9356_CSM_RESET, 0x00);
+ regmap_write(es9356->regmap, ES9356_HP_TYPE, 0x08);
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_L), ES9356_DEFAULT_VOLUME);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_FU113, ES9356_SDCA_CTL_FU_VOLUME, CH_R), ES9356_DEFAULT_VOLUME);
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_L), ES9356_DEFAULT_VOLUME);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_FU21, ES9356_SDCA_CTL_FU_VOLUME, CH_R), ES9356_DEFAULT_VOLUME);
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_L), ES9356_DEFAULT_VOLUME);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU41, ES9356_SDCA_CTL_FU_VOLUME, CH_R), ES9356_DEFAULT_VOLUME);
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_L), ES9356_DEFAULT_VOLUME);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_FU36, ES9356_SDCA_CTL_FU_VOLUME, CH_R), ES9356_DEFAULT_VOLUME);
+}
+
+static int es9356_sdca_io_init(struct device *dev, struct sdw_slave *slave)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(&slave->dev);
+
+ if (es9356->hw_init)
+ return 0;
+
+ es9356->disable_irq = false;
+
+ regcache_cache_only(es9356->regmap, false);
+
+ if (es9356->first_hw_init) {
+ regcache_cache_bypass(es9356->regmap, true);
+ } else {
+ /*
+ * PM runtime is only enabled when a Slave reports as Attached
+ */
+
+ /* set autosuspend parameters */
+ pm_runtime_set_autosuspend_delay(&slave->dev, 3000);
+ pm_runtime_use_autosuspend(&slave->dev);
+
+ /* update count of parent 'active' children */
+ pm_runtime_set_active(&slave->dev);
+
+ /* make sure the device does not suspend immediately */
+ pm_runtime_mark_last_busy(&slave->dev);
+
+ pm_runtime_enable(&slave->dev);
+
+ es9356_register_init(es9356);
+ }
+ pm_runtime_get_noresume(&slave->dev);
+
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT_XU12, ES9356_SDCA_CTL_SELECTED_MODE, 0), 0x01);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_MIC, ES9356_SDCA_ENT0, ES9356_SDCA_CTL_FUNC_STATUS, 0), 0x40);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT_XU22, ES9356_SDCA_CTL_SELECTED_MODE, 0), 0x01);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_AMP, ES9356_SDCA_ENT0, ES9356_SDCA_CTL_FUNC_STATUS, 0), 0x40);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_XU42, ES9356_SDCA_CTL_SELECTED_MODE, 0), 0x01);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT_XU36, ES9356_SDCA_CTL_SELECTED_MODE, 0), 0x01);
+ regmap_write(es9356->regmap,
+ SDW_SDCA_CTL(FUNC_NUM_UAJ, ES9356_SDCA_ENT0, ES9356_SDCA_CTL_FUNC_STATUS, 0), 0x40);
+
+ if (es9356->first_hw_init) {
+ regcache_cache_bypass(es9356->regmap, false);
+ regcache_mark_dirty(es9356->regmap);
+ } else
+ es9356->first_hw_init = true;
+
+ es9356->hw_init = true;
+
+ pm_runtime_mark_last_busy(&slave->dev);
+ pm_runtime_put_autosuspend(&slave->dev);
+
+ return 0;
+}
+
+static int es9356_sdw_update_status(struct sdw_slave *slave,
+ enum sdw_slave_status status)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(&slave->dev);
+
+ if (status == SDW_SLAVE_UNATTACHED) {
+ es9356->hw_init = false;
+ cancel_delayed_work_sync(&es9356->interrupt_handle_work);
+ cancel_delayed_work_sync(&es9356->button_detect_work);
+ regcache_cache_only(es9356->regmap, true);
+ }
+
+ if (status == SDW_SLAVE_ATTACHED) {
+ if (es9356->hs_jack)
+ sdw_write_no_pm(es9356->slave, SDW_SCP_SDCA_INTMASK1,
+ (SDW_SCP_SDCA_INTMASK_SDCA_7 | SDW_SCP_SDCA_INTMASK_SDCA_5 | SDW_SCP_SDCA_INTMASK_SDCA_1));
+ }
+
+ if (es9356->hw_init || status != SDW_SLAVE_ATTACHED)
+ return 0;
+
+ return es9356_sdca_io_init(&slave->dev, slave);
+}
+
+static int es9356_sdw_read_prop(struct sdw_slave *slave)
+{
+ struct sdw_slave_prop *prop = &slave->prop;
+ int nval;
+ int i, j;
+ u32 bit;
+ unsigned long addr;
+ struct sdw_dpn_prop *dpn;
+
+ prop->paging_support = true;
+
+ /*
+ * first we need to allocate memory for set bits in port lists
+ * the port allocation is completely arbitrary:
+ * DP0 is not supported
+ * DP3 and DP4 is sink
+ * DP1 and DP2 is source
+ */
+ prop->source_ports = BIT(1) | BIT(2);
+ prop->sink_ports = BIT(3) | BIT(4);
+
+ nval = hweight32(prop->source_ports);
+ prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval,
+ sizeof(*prop->src_dpn_prop),
+ GFP_KERNEL);
+ if (!prop->src_dpn_prop)
+ return -ENOMEM;
+
+ i = 0;
+ dpn = prop->src_dpn_prop;
+ addr = prop->source_ports;
+ for_each_set_bit(bit, &addr, 32) {
+ dpn[i].num = bit;
+ dpn[i].type = SDW_DPN_FULL;
+ dpn[i].simple_ch_prep_sm = true;
+ i++;
+ }
+
+ /* do this again for sink now */
+ nval = hweight32(prop->sink_ports);
+ prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval,
+ sizeof(*prop->sink_dpn_prop),
+ GFP_KERNEL);
+ if (!prop->sink_dpn_prop)
+ return -ENOMEM;
+
+ j = 0;
+ dpn = prop->sink_dpn_prop;
+ addr = prop->sink_ports;
+ for_each_set_bit(bit, &addr, 32) {
+ dpn[j].num = bit;
+ dpn[j].type = SDW_DPN_FULL;
+ dpn[j].simple_ch_prep_sm = true;
+ j++;
+ }
+
+ /* wake-up event */
+ prop->wake_capable = 1;
+
+ return 0;
+}
+
+static int es9356_sdw_interrupt_callback(struct sdw_slave *slave,
+ struct sdw_slave_intr_status *status)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(&slave->dev);
+ int ret, stat, reg;
+ int count = 0, retry = 3;
+ unsigned int sdca_cascade, scp_sdca_stat1 = 0;
+
+ mutex_lock(&es9356->jack_lock);
+
+ ret = sdw_read_no_pm(es9356->slave, SDW_SCP_SDCA_INT1);
+ if (ret < 0)
+ goto io_error;
+ es9356->sdca_status = ret;
+
+ do {
+ reg = sdw_read_no_pm(es9356->slave, SDW_SCP_SDCA_INT1);
+ if (reg < 0)
+ goto io_error;
+ if (reg & SDW_SCP_SDCA_INTMASK_SDCA_1) {
+ ret = sdw_update_no_pm(es9356->slave, SDW_SCP_SDCA_INT1,
+ SDW_SCP_SDCA_INT_SDCA_1, SDW_SCP_SDCA_INT_SDCA_1);
+ if (ret < 0)
+ goto io_error;
+ }
+
+ if (reg & SDW_SCP_SDCA_INTMASK_SDCA_5) {
+ ret = sdw_update_no_pm(es9356->slave, SDW_SCP_SDCA_INT1,
+ SDW_SCP_SDCA_INT_SDCA_5, SDW_SCP_SDCA_INT_SDCA_5);
+ if (ret < 0)
+ goto io_error;
+ }
+
+ if (reg & SDW_SCP_SDCA_INTMASK_SDCA_7) {
+ ret = sdw_update_no_pm(es9356->slave, SDW_SCP_SDCA_INT1,
+ SDW_SCP_SDCA_INT_SDCA_7, SDW_SCP_SDCA_INT_SDCA_7);
+ if (ret < 0)
+ goto io_error;
+ }
+
+ ret = sdw_read_no_pm(es9356->slave, SDW_DP0_INT);
+ if (ret < 0)
+ goto io_error;
+ sdca_cascade = ret & SDW_DP0_SDCA_CASCADE;
+
+ ret = sdw_read_no_pm(es9356->slave, SDW_SCP_SDCA_INT1);
+ if (ret < 0)
+ goto io_error;
+ scp_sdca_stat1 = ret &
+ (SDW_SCP_SDCA_INTMASK_SDCA_1 | SDW_SCP_SDCA_INTMASK_SDCA_5 | SDW_SCP_SDCA_INTMASK_SDCA_7);
+
+ stat = scp_sdca_stat1 || sdca_cascade;
+
+ count++;
+ } while (stat != 0 && count < retry);
+
+ /* The 280 ms figure was determined through testing */
+ if (status->sdca_cascade && !es9356->disable_irq)
+ mod_delayed_work(system_power_efficient_wq,
+ &es9356->interrupt_handle_work, msecs_to_jiffies(280));
+
+ mutex_unlock(&es9356->jack_lock);
+ return 0;
+
+io_error:
+ mutex_unlock(&es9356->jack_lock);
+ pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret);
+ return ret;
+}
+
+static const struct sdw_slave_ops es9356_sdw_slave_ops = {
+ .read_prop = es9356_sdw_read_prop,
+ .interrupt_callback = es9356_sdw_interrupt_callback,
+ .update_status = es9356_sdw_update_status,
+};
+
+static int es9356_sdca_init(struct device *dev, struct regmap *regmap, struct sdw_slave *slave)
+{
+ struct es9356_sdw_priv *es9356;
+
+ es9356 = devm_kzalloc(dev, sizeof(*es9356), GFP_KERNEL);
+ if (!es9356)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, es9356);
+
+ es9356->slave = slave;
+ es9356->regmap = regmap;
+ mutex_init(&es9356->jack_lock);
+ mutex_init(&es9356->pde_lock);
+
+ regcache_cache_only(es9356->regmap, true);
+
+ es9356->hw_init = false;
+ es9356->first_hw_init = false;
+
+ INIT_DELAYED_WORK(&es9356->interrupt_handle_work,
+ es9356_interrupt_handler);
+ INIT_DELAYED_WORK(&es9356->button_detect_work,
+ es9356_button_detect_handler);
+
+ return devm_snd_soc_register_component(dev,
+ &snd_soc_es9356_sdw_component,
+ es9356_sdw_dai,
+ ARRAY_SIZE(es9356_sdw_dai));
+}
+
+static int es9356_sdw_probe(struct sdw_slave *slave,
+ const struct sdw_device_id *id)
+{
+ struct regmap *regmap;
+
+ regmap = devm_regmap_init_sdw_mbq_cfg(&slave->dev, slave, &es9356_sdca_regmap, &es9356_mbq_config);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return es9356_sdca_init(&slave->dev, regmap, slave);
+}
+
+static int es9356_sdw_remove(struct sdw_slave *slave)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(&slave->dev);
+
+ if (es9356->hw_init) {
+ cancel_delayed_work_sync(&es9356->interrupt_handle_work);
+ cancel_delayed_work_sync(&es9356->button_detect_work);
+ }
+
+ if (es9356->first_hw_init)
+ pm_runtime_disable(&slave->dev);
+
+ mutex_destroy(&es9356->jack_lock);
+ mutex_destroy(&es9356->pde_lock);
+
+ return 0;
+}
+
+static const struct sdw_device_id es9356_sdw_id[] = {
+ SDW_SLAVE_ENTRY_EXT(0x04b3, 0x9356, 0x02, 0, 0),
+ SDW_SLAVE_ENTRY_EXT(0x04b3, 0x9356, 0x03, 0, 0),
+ {},
+};
+MODULE_DEVICE_TABLE(sdw, es9356_sdw_id);
+
+static int es9356_sdca_dev_suspend(struct device *dev)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(dev);
+
+ mutex_lock(&es9356->jack_lock);
+ es9356->disable_irq = true;
+ cancel_delayed_work_sync(&es9356->interrupt_handle_work);
+ cancel_delayed_work_sync(&es9356->button_detect_work);
+ mutex_unlock(&es9356->jack_lock);
+
+ return 0;
+}
+
+static int es9356_sdca_dev_system_suspend(struct device *dev)
+{
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(dev);
+
+ return es9356_sdca_dev_suspend(dev);
+}
+
+#define es9356_PROBE_TIMEOUT 2000
+
+static int es9356_sdca_dev_resume(struct device *dev)
+{
+ struct sdw_slave *slave = dev_to_sdw_dev(dev);
+ struct es9356_sdw_priv *es9356 = dev_get_drvdata(dev);
+ unsigned long time;
+
+ if (!slave->unattach_request) {
+ es9356->disable_irq = false;
+ goto regmap_sync;
+ }
+
+ time = wait_for_completion_timeout(&slave->initialization_complete,
+ msecs_to_jiffies(es9356_PROBE_TIMEOUT));
+ if (!time) {
+ dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ sdw_show_ping_status(slave->bus, true);
+
+ return -ETIMEDOUT;
+ }
+
+regmap_sync:
+ slave->unattach_request = 0;
+ regcache_cache_only(es9356->regmap, false);
+ regcache_sync(es9356->regmap);
+ return 0;
+}
+
+static const struct dev_pm_ops es9356_sdca_pm = {
+ SYSTEM_SLEEP_PM_OPS(es9356_sdca_dev_system_suspend, es9356_sdca_dev_resume)
+ RUNTIME_PM_OPS(es9356_sdca_dev_suspend, es9356_sdca_dev_resume, NULL)
+};
+
+static struct sdw_driver es9356_sdw_driver = {
+ .driver = {
+ .name = "es9356",
+ .pm = pm_ptr(&es9356_sdca_pm),
+ },
+ .probe = es9356_sdw_probe,
+ .remove = es9356_sdw_remove,
+ .ops = &es9356_sdw_slave_ops,
+ .id_table = es9356_sdw_id,
+};
+module_sdw_driver(es9356_sdw_driver);
+
+MODULE_IMPORT_NS("SND_SOC_SDCA");
+MODULE_DESCRIPTION("ASoC ES9356 SDCA SDW codec driver");
+MODULE_AUTHOR("Michael Zhang <zhangyi@everest-semi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es9356.h b/sound/soc/codecs/es9356.h
new file mode 100644
index 000000000..2a676f365
--- /dev/null
+++ b/sound/soc/codecs/es9356.h
@@ -0,0 +1,208 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+
+#ifndef __ES9356_H__
+#define __ES9356_H__
+
+/*ES9356 Implementation-define*/
+#define ES9356_FLAGS_HP 0x2003
+#define ES9356_CSM_RESET 0x2020
+#define ES9356_FUC_RESET 0x2021
+#define ES9356_STATE 0x2022
+#define ES9356_VMID_TIME 0x2023
+#define ES9356_STATE_TIME 0x2024
+#define ES9356_HP_SPK_TIME 0x2025
+#define ES9356_WP_ENABLE 0x2026
+#define ES9356_DMIC_GPIO 0x2027
+#define ES9356_ENDPOINT_MODE 0x2028
+
+/*HP DETECT*/
+#define ES9356_HP_TYPE 0x2029
+#define ES9356_HP_DETECTTIME 0x202A
+#define ES9356_MICBIAS_SEL 0x202B
+#define ES9356_KEY_PRESS_TIME 0x202C
+#define ES9356_KEY_RELEASE_TIME 0x202D
+#define ES9356_KEY_HOLD_TIME 0x202E
+#define ES9356_BTSEL_REF 0x202F
+#define ES9356_BUTTON_CHARGE 0x2030
+
+#define ES9356_KEYD_DETECT 0x2031
+#define ES9356_DPEN_TIME 0x2032
+#define ES9356_TIMER_CHECK 0x2033
+#define ES9356_IBIASGEN 0x2041
+#define ES9356_VMID1SEL 0x2042
+#define ES9356_VMID1STL 0x2043
+#define ES9356_VMID2SEL 0x2044
+#define ES9356_VMID2STL 0x2045
+#define ES9356_VSEL 0x2046
+#define ES9356_MICBIAS_CTL 0x2047
+#define ES9356_HPDETECT_CTL 0x2048
+#define ES9356_MICBIAS_RES 0x2049
+
+/*CLK*/
+#define ES9356_CLK_SEL 0x2050
+#define ES9356_CLK_CTL 0x2051
+#define ES9356_DETCLK_CTL 0x2052
+#define ES9356_CPCLK_CTL 0x2053
+#define ES9356_SPKCLK_CTL 0x2054
+#define ES9356_PRE_DIV_CTL 0x2055
+#define ES9356_DLL_MODE 0x2056
+#define ES9356_ANACLK_SEL 0x2057
+#define ES9356_OSRCLK_SEL 0x2058
+#define ES9356_DSPCLK_SEL 0x2059
+#define ES9356_SPK9M_MODE 0x205a
+
+/*ADC DIG CTL*/
+#define ES9356_DMIC_POL 0x2061
+#define ES9356_ADC_SWAP 0x2062
+#define ES9356_ADC_OSR 0x2063
+#define ES9356_ADC_OSRGAIN 0x2064
+#define ES9356_ADC_CLEARRAM 0x2065
+#define ES9356_ADC_RAMP 0x2066
+#define ES9356_ADC_HPF1 0x2067
+#define ES9356_ADC_HPF2 0x2068
+#define ES9356_ADC_ALC 0x206C
+#define ES9356_ALC_LEVEL 0x206D
+#define ES9356_ALC_RAMP_WINSIZE 0x206E
+
+/*ADC ANA CTL*/
+#define ES9356_ADC_REF_EN 0x2080
+#define ES9356_ADC_AMIC_CTL 0x2081
+#define ES9356_ADC_ANA 0x2082
+#define ES9356_PGA_CTL 0x2083
+#define ES9356_ADC_INT 0x2084
+#define ES9356_ADC_VCM 0x2085
+#define ES9356_ADC_VRPBIAS 0x2086
+#define ES9356_ADC_LP 0x2087
+
+/*DAC DIG CTL*/
+#define ES9356_DAC_FSMODE 0x2090
+#define ES9356_DAC_OSR 0x2091
+#define ES9356_DAC_INV 0x2092
+#define ES9356_DAC_RAMP 0x2093
+#define ES9356_DAC_VPPSCALE 0x2094
+#define ES9356_DAC_SWAP 0x2097
+#define ES9356_SPKCMP_VPPSC 0x20A0
+#define ES9356_CALIBRATION_TIME 0x20A1
+#define ES9356_CALIBRATION_SETTING 0x20A2
+#define ES9356_DAC_OFFSET_LH 0x20A3
+#define ES9356_DAC_OFFSET_LL 0x20A4
+#define ES9356_DAC_OFFSET_RH 0x20A5
+#define ES9356_DAC_OFFSET_RL 0x20A6
+
+/*DAC ANA CTL*/
+#define ES9356_DAC_REF_EN 0x20B0
+#define ES9356_DAC_ENABLE 0x20B1
+#define ES9356_DAC_VROI 0x20B2
+#define ES9356_DAC_LP 0x20B3
+
+/*HP CTL*/
+#define ES9356_CHARGEPUMP_CTL 0x20C0
+#define ES9356_CPLDO_CTL 0x20C1
+#define ES9356_HP_REF_CTL 0x20C2
+#define ES9356_HP_IBIAS 0x20C3
+#define ES9356_HP_EN 0x20C4
+#define ES9356_HP_VOLUME 0x20C5
+#define ES9356_HP_LP 0x20C6
+
+/*SPK CTL*/
+#define ES9356_SPKLDO_CTL 0x20D0
+#define ES9356_CLASSD_CTL 0x20D1
+#define ES9356_SPK_HBDG 0x20D5
+#define ES9356_SPK_VOLUME 0x20D7
+#define ES9356_SPK_SCP 0x20D8
+#define ES9356_SPK_DT 0x20D9
+#define ES9356_SPK_OTP 0x20DA
+#define ES9356_SPKBIAS_COMP 0x20DB
+
+/* ES9356 SDCA Control - function number */
+#define FUNC_NUM_UAJ 0x01
+#define FUNC_NUM_MIC 0x02
+#define FUNC_NUM_AMP 0x03
+#define FUNC_NUM_HID 0x04
+
+/* ES9356 SDCA entity */
+#define ES9356_SDCA_ENT0 0x00
+#define ES9356_SDCA_ENT_PDE11 0x03
+#define ES9356_SDCA_ENT_FU11 0x04
+#define ES9356_SDCA_ENT_XU12 0x05
+#define ES9356_SDCA_ENT_FU113 0x07
+#define ES9356_SDCA_ENT_CS113 0x09
+#define ES9356_SDCA_ENT_PPU11 0x0C
+
+#define ES9356_SDCA_ENT_CS21 0x02
+#define ES9356_SDCA_ENT_PPU21 0x03
+#define ES9356_SDCA_ENT_FU21 0X04
+#define ES9356_SDCA_ENT_XU22 0x06
+#define ES9356_SDCA_ENT_SAPU29 0x03
+#define ES9356_SDCA_ENT_PDE23 0x0B
+#define ES9356_SDCA_ENT_HID01 0x01
+
+#define ES9356_SDCA_ENT_CS41 0x02
+#define ES9356_SDCA_ENT_FU35 0x04
+#define ES9356_SDCA_ENT_XU42 0x06
+#define ES9356_SDCA_ENT_FU41 0x07
+#define ES9356_SDCA_ENT_PDE47 0x0E
+#define ES9356_SDCA_ENT_IT33 0x0F
+#define ES9356_SDCA_ENT_PDE34 0x10
+#define ES9356_SDCA_ENT_FU33 0x11
+#define ES9356_SDCA_ENT_XU36 0x13
+#define ES9356_SDCA_ENT_FU36 0x15
+#define ES9356_SDCA_ENT_CS36 0x17
+#define ES9356_SDCA_ENT_GE35 0x18
+
+/* ES9356 SDCA control */
+#define ES9356_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10
+#define ES9356_SDCA_CTL_FU_MUTE 0x01
+#define ES9356_SDCA_CTL_FU_VOLUME 0x02
+#define ES9356_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10
+#define ES9356_SDCA_CTL_SELECTED_MODE 0x01
+#define ES9356_SDCA_CTL_DETECTED_MODE 0x02
+#define ES9356_SDCA_CTL_REQ_POWER_STATE 0x01
+#define ES9356_SDCA_CTL_FU_CH_GAIN 0x0b
+#define ES9356_SDCA_CTL_FUNC_STATUS 0x10
+#define ES9356_SDCA_CTL_ACTUAL_POWER_STATE 0x10
+#define ES9356_SDCA_CTL_POSTURE_NUMBER 0x00
+
+/* ES9356 SDCA channel */
+#define CH_L 0x01
+#define CH_R 0x02
+#define MBQ 0x2000
+
+/* ES9356 HID*/
+#define ES9356_BUF_ADDR_HID 0x44000000
+#define ES9356_HID_BYTE2 0x44000001
+#define ES9356_HID_BYTE3 0x44000002
+#define ES9356_HID_BYTE4 0x44000003
+
+/* ES9356 Volume Setting*/
+#define ES9356_VU_BASE 768
+#define ES9356_OFFSET_HIGH 0x07F8
+#define ES9356_OFFSET_LOW 0x0007
+#define ES9356_DEFAULT_VOLUME 0x00
+#define ES9356_VOLUME_STEP 32
+#define ES9356_VOLUME_MIN -768
+#define ES9356_VOLUME_MAX 285
+#define ES9356_AMIC_GAIN_STEP 768
+#define ES9356_DMIC_GAIN_STEP 1536
+#define ES9356_GAIN_MIN 0
+#define ES9356_AMIC_GAIN_MAX 10
+#define ES9356_DMIC_GAIN_MAX 3
+
+enum {
+ ES9356_DMIC = 1, /* For dmic */
+ ES9356_JACK_IN, /* For headset mic */
+ ES9356_AMP, /* For speaker */
+ ES9356_JACK_OUT, /* For headphone */
+};
+
+enum {
+ ES9356_SDCA_RATE_16000HZ,
+ ES9356_SDCA_RATE_24000HZ,
+ ES9356_SDCA_RATE_32000HZ,
+ ES9356_SDCA_RATE_44100HZ,
+ ES9356_SDCA_RATE_48000HZ,
+ ES9356_SDCA_RATE_88200HZ,
+ ES9356_SDCA_RATE_96000HZ,
+};
+
+#endif
--
2.17.1
^ permalink raw reply related
* [PATCH v9 2/5] ASoC: sdw_utils: add ES9356 in codec_info_list
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
In-Reply-To: <20260420094349.14197-1-zhangyi@everest-semi.com>
Add ES9356 in codec_info_list
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
---
sound/soc/sdw_utils/soc_sdw_utils.c | 50 +++++++++++++++++++++++++++++
1 file changed, 50 insertions(+)
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index 2807f536e..840242d7f 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -938,6 +938,56 @@ struct asoc_sdw_codec_info codec_info_list[] = {
},
.aux_num = 1,
},
+ {
+ .vendor_id = 0x04b3,
+ .part_id = 0x9356,
+ .name_prefix = "es9356",
+ .version_id = 3,
+ .dais = {
+ {
+ .direction = {true, false},
+ .dai_name = "es9356-sdp-aif1",
+ .dai_type = SOC_SDW_DAI_TYPE_JACK,
+ .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
+ .init = asoc_sdw_es9356_init,
+ .exit = asoc_sdw_es9356_exit,
+ .rtd_init = asoc_sdw_es9356_rtd_init,
+ .controls = generic_jack_controls,
+ .num_controls = ARRAY_SIZE(generic_jack_controls),
+ .widgets = generic_jack_widgets,
+ .num_widgets = ARRAY_SIZE(generic_jack_widgets),
+ },
+ {
+ .direction = {false, true},
+ .dai_name = "es9356-sdp-aif4",
+ .dai_type = SOC_SDW_DAI_TYPE_MIC,
+ .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID},
+ .rtd_init = asoc_sdw_es9356_dmic_rtd_init,
+ .widgets = generic_dmic_widgets,
+ .num_widgets = ARRAY_SIZE(generic_dmic_widgets),
+ },
+ {
+ .direction = {false, true},
+ .dai_name = "es9356-sdp-aif2",
+ .dai_type = SOC_SDW_DAI_TYPE_JACK,
+ .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_JACK_IN_DAI_ID},
+ },
+ {
+ .direction = {true, false},
+ .dai_name = "es9356-sdp-aif3",
+ .component_name = "es9356",
+ .dai_type = SOC_SDW_DAI_TYPE_AMP,
+ .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID},
+ .init = asoc_sdw_es9356_amp_init,
+ .rtd_init = asoc_sdw_es9356_spk_rtd_init,
+ .controls = generic_spk_controls,
+ .num_controls = ARRAY_SIZE(generic_spk_controls),
+ .widgets = generic_spk_widgets,
+ .num_widgets = ARRAY_SIZE(generic_spk_widgets),
+ },
+ },
+ .dai_num = 4,
+ },
{
.vendor_id = 0x0105,
.part_id = 0xaaaa, /* generic codec mockup */
--
2.17.1
^ permalink raw reply related
* [PATCH v9 0/5] Add es9356 focused SoundWire CODEC
From: Zhang Yi @ 2026-04-20 9:43 UTC (permalink / raw)
To: broonie, tiwai, linux-sound
Cc: peter.ujfalusi, yung-chuan.liao, ranjani.sridharan, kai.vehmanen,
ckeepax, Zhang Yi
This patch chain adds support for the ES9356 PC focused
SoundWire CODEC.
v9 -> v8:
-Fix some formatting issues in es9356.c
-Removed regmap_write(SDW_SCP_SYSTEMCTRL) from the codec driver
v8 -> v7:
-When the device status is UNATTACHED,
set regmap to cache-only and cancel any delay work already in the queue.
v7 -> v6:
-Provide comments for some parameters
-Rename `jack_detect_work` to `interrupt_handle_work`
-Add a jack_lock when handling disable_irq in suspend
-Adjust the handling procedure when a jack register
is incorrectly identified
-Replace the custom SOC_SINGLE_Q78_TLV with SDCA_SINGLE_Q78_TLV
v6 -> v5:
-Handling default values in `button_detect` and `jack_detect`
-Add the member `disable_irq` so that interrupts are
no longer processed after suspend.
-Move the PDE processing to `hw_params` and `hw_free`
-Separate JACK_OUT and JACK_IN in hw_params
-Simplify reading and writing to SDW_SCP_SDCA_INT1 in the callback
-Adjust the scope of `regcache_sync` in the resume
v5 -> v4:
-Merge patch1 and patch3
-Add vendor_id in codec_info
-Use q78_get and q78_set to create Kcontrol
-Set mbq_size and use regmap_mbq
-An additional helper function to implement DAPM
v4 -> v3:
-Updated some KControl names
-add copyright in soc_sdw_es9356
v3 -> v2:
-Removed unnecessary section in soc_sdw_es9356.c
-Add `pm_runtime_put_noidle` to `es9356.c` to
reduce device incremented count
-Updated some KControl names
-Fix the error related to jack_report in the codec driver
v2 -> v1:
-Adjusted mutexes in the ES9356 driver
-Use macros to replace certain values
-Adjusted some ordering issues in Kconfig.
v1 -> v0:
-Modified snd_soc_dapm_add_routes in soc_sdw_es9356
Zhang Yi (5):
ASoC: sdw_utils: add soc_sdw_es9356
ASoC: sdw_utils: add ES9356 in codec_info_list
ASoC: es9356-sdca: Add ES9356 SDCA driver
ASoC: Intel: soc-acpi: arl: Add es9356 support
ASoC: Intel: sof_sdw: add es9356 support
include/sound/soc_sdw_utils.h | 14 +
sound/soc/codecs/Kconfig | 7 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/es9356.c | 1154 +++++++++++++++++
sound/soc/codecs/es9356.h | 208 +++
sound/soc/intel/boards/Kconfig | 1 +
.../intel/common/soc-acpi-intel-arl-match.c | 52 +
sound/soc/sdw_utils/Makefile | 1 +
sound/soc/sdw_utils/soc_sdw_es9356.c | 230 ++++
sound/soc/sdw_utils/soc_sdw_utils.c | 50 +
10 files changed, 1719 insertions(+)
create mode 100644 sound/soc/codecs/es9356.c
create mode 100644 sound/soc/codecs/es9356.h
create mode 100644 sound/soc/sdw_utils/soc_sdw_es9356.c
--
2.17.1
^ permalink raw reply
* Re: [PATCH v1 0/2] firewire: Simplify storing pointers in device id struct
From: Takashi Sakamoto @ 2026-04-20 9:08 UTC (permalink / raw)
To: Uwe Kleine-König (The Capable Hub)
Cc: Clemens Ladisch, Jaroslav Kysela, Takashi Iwai,
Christian A. Ehrhardt, Christian A. Ehrhardt, linux1394-devel,
linux-sound, Wolfram Sang, Andy Shevchenko
In-Reply-To: <cover.1776579304.git.u.kleine-koenig@baylibre.com>
Hi,
Thanks for the patches. As far as I can see, they can be applied neither
any compilation failures and running regressions.
We are in the middle of merge window for v7.2. I had not planned to send any
changes to upstream for firewire subsystem, but there is still some time
before it closes. If the sound subsystem maintainer does not mind, I
would like to proceed.
Just out of curiosity, what does the CHERI extension adopted to RISC-V
architecture require in terms of kernel programming? Is taking extra
care when storing pointer values in long-type variables sufficient in
driver code?
Thanks
Takashi Sakamoto
On Sun, Apr 19, 2026 at 08:42:12AM +0200, Uwe Kleine-König (The Capable Hub) wrote:
> Hello,
>
> <linux/mod_devicetable.h> contains several device_id structs for various
> device types.
>
> Most of them have one of:
>
> - kernel_ulong_t driver_data (sometimes called "driver_info", sometimes
> the type is plain unsigned long)
> - const void *data (sometimes called "driver_data" or "context", sometimes not const)
>
> A considerable amount of drivers for the first category uses the
> unsigned long variable to store a pointer. This involves casting both
> for assignment and usage.
>
> An additional complication exists for the CHERI hardware extension
> where sizeof(void *) > sizeof(unsigned long). So with that an unsigned
> long variable cannot be used to store a pointer.
>
> To address both issues this series replaces the unsigned long variable
> by an anonymous union containing both an unsigned long and a pointer.
>
> For all non-CHERI architectures this isn't an ABI change because all
> have sizeof(void *) == sizeof(unsigned long).
>
> The first patch changes the definition of struct ieee1394_device_id. The
> second drops some casts in sound drivers. (There are no other firewire
> drivers that could benefit.) I adapted all sound drivers in a single
> patch, tell me if I should split per driver.
>
> For merging I suggest to take the whole series via the ALSA tree in the
> next merge window, as there are no modified files that are specific to
> firewire only and the second patch depends on the first.
>
> Best regards
> Uwe
>
> Uwe Kleine-König (The Capable Hub) (2):
> firewire: Simplify storing pointers in device id struct
> ALSA: firewire: Make use of ieee1394's .driver_data_ptr
>
> include/linux/mod_devicetable.h | 5 ++++-
> sound/firewire/dice/dice.c | 34 ++++++++++++++++-----------------
> sound/firewire/fireface/ff.c | 12 ++++++------
> sound/firewire/motu/motu.c | 6 +++---
> sound/firewire/oxfw/oxfw.c | 4 ++--
> 5 files changed, 32 insertions(+), 29 deletions(-)
>
>
> base-commit: 028ef9c96e96197026887c0f092424679298aae8
> --
> 2.47.3
>
^ permalink raw reply
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