date:20240620

qmc_chan_get_byphandle() and the resource managed version retrieve a
channel from a simple phandle.

Extend the API and introduce qmc_chan_get_byphandles_index() and the
resource managed version in order to retrieve a channel from a phandle
list using the provided index to identify the phandle in the list.

Also update qmc_chan_get_byphandle() and the resource managed version to
use qmc_chan_get_byphandles_index() and so avoid code duplication.

Signed-off-by: Herve Codina 
---
 drivers/soc/fsl/qe/qmc.c | 19 +++
 include/soc/fsl/qe/qmc.h | 25 ++---
 2 files changed, 33 insertions(+), 11 deletions(-)

diff --git a/drivers/soc/fsl/qe/qmc.c b/drivers/soc/fsl/qe/qmc.c
index f498db9abe35..e23d60018400 100644
--- a/drivers/soc/fsl/qe/qmc.c
+++ b/drivers/soc/fsl/qe/qmc.c
@@ -1777,13 +1777,15 @@ static struct qmc_chan *qmc_chan_get_from_qmc(struct 
device_node *qmc_np, unsign
return qmc_chan;
 }
 
-struct qmc_chan *qmc_chan_get_byphandle(struct device_node *np, const char 
*phandle_name)
+struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np,
+  const char *phandles_name,
+  int index)
 {
struct of_phandle_args out_args;
struct qmc_chan *qmc_chan;
int ret;
 
-   ret = of_parse_phandle_with_fixed_args(np, phandle_name, 1, 0,
+   ret = of_parse_phandle_with_fixed_args(np, phandles_name, 1, index,
   &out_args);
if (ret < 0)
return ERR_PTR(ret);
@@ -1797,7 +1799,7 @@ struct qmc_chan *qmc_chan_get_byphandle(struct 
device_node *np, const char *phan
of_node_put(out_args.np);
return qmc_chan;
 }
-EXPORT_SYMBOL(qmc_chan_get_byphandle);
+EXPORT_SYMBOL(qmc_chan_get_byphandles_index);
 
 struct qmc_chan *qmc_chan_get_bychild(struct device_node *np)
 {
@@ -1827,9 +1829,10 @@ static void devm_qmc_chan_release(struct device *dev, 
void *res)
qmc_chan_put(*qmc_chan);
 }
 
-struct qmc_chan *devm_qmc_chan_get_byphandle(struct device *dev,
-struct device_node *np,
-const char *phandle_name)
+struct qmc_chan *devm_qmc_chan_get_byphandles_index(struct device *dev,
+   struct device_node *np,
+   const char *phandles_name,
+   int index)
 {
struct qmc_chan *qmc_chan;
struct qmc_chan **dr;
@@ -1838,7 +1841,7 @@ struct qmc_chan *devm_qmc_chan_get_byphandle(struct 
device *dev,
if (!dr)
return ERR_PTR(-ENOMEM);
 
-   qmc_chan = qmc_chan_get_byphandle(np, phandle_name);
+   qmc_chan = qmc_chan_get_byphandles_index(np, phandles_name, index);
if (!IS_ERR(qmc_chan)) {
*dr = qmc_chan;
devres_add(dev, dr);
@@ -1848,7 +1851,7 @@ struct qmc_chan *devm_qmc_chan_get_byphandle(struct 
device *dev,
 
return qmc_chan;
 }
-EXPORT_SYMBOL(devm_qmc_chan_get_byphandle);
+EXPORT_SYMBOL(devm_qmc_chan_get_byphandles_index);
 
 struct qmc_chan *devm_qmc_chan_get_bychild(struct device *dev,
   struct device_node *np)
diff --git a/include/soc/fsl/qe/qmc.h b/include/soc/fsl/qe/qmc.h
index 2a333fc1ea81..0fa7205145ce 100644
--- a/include/soc/fsl/qe/qmc.h
+++ b/include/soc/fsl/qe/qmc.h
@@ -16,11 +16,30 @@ struct device_node;
 struct device;
 struct qmc_chan;
 
-struct qmc_chan *qmc_chan_get_byphandle(struct device_node *np, const char 
*phandle_name);
+struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np,
+  const char *phandles_name,
+  int index);
+struct qmc_chan *devm_qmc_chan_get_byphandles_index(struct device *dev,
+   struct device_node *np,
+   const char *phandles_name,
+   int index);
+
+static inline struct qmc_chan *qmc_chan_get_byphandle(struct device_node *np,
+ const char *phandle_name)
+{
+   return qmc_chan_get_byphandles_index(np, phandle_name, 0);
+}
+
+static inline struct qmc_chan *devm_qmc_chan_get_byphandle(struct device *dev,
+  struct device_node 
*np,
+  const char 
*phandle_name)
+{
+   return devm_qmc_chan_get_byphandles_index(dev, np, phandle_name, 0);
+}
+
 struct qmc_chan *qmc_chan_get_bychild(struct device_node *np);
 void qmc_chan_put(struct qmc_chan *chan);
-struct qmc_chan *devm_qmc_chan_get_byphandle(struct device *dev, struct 
device_node *np,
-

[PATCH 06/10] ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved()

Constraints are set by qmc_dai_startup(). These constraints are specific
to the interleaved mode.

With the future introduction of support for non-interleaved mode, a new
set of constraints will be set. To make the code clear and keep
qmc_dai_startup() simple, extract the current interleaved mode
constraints settings to a specific function.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 37 +--
 1 file changed, 22 insertions(+), 15 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index 36145f1ddbf1..f70c6c8eec4a 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -436,24 +436,14 @@ static int 
qmc_dai_hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *
return qmc_dai_hw_rule_format_by_channels(qmc_dai, params, 
qmc_dai->nb_rx_ts);
 }
 
-static int qmc_dai_startup(struct snd_pcm_substream *substream,
-  struct snd_soc_dai *dai)
+static int qmc_dai_constraints_interleaved(struct snd_pcm_substream *substream,
+  struct qmc_dai *qmc_dai)
 {
-   struct qmc_dai_prtd *prtd = substream->runtime->private_data;
snd_pcm_hw_rule_func_t hw_rule_channels_by_format;
snd_pcm_hw_rule_func_t hw_rule_format_by_channels;
-   struct qmc_dai *qmc_dai;
unsigned int frame_bits;
int ret;
 
-   qmc_dai = qmc_dai_get_data(dai);
-   if (!qmc_dai) {
-   dev_err(dai->dev, "Invalid dai\n");
-   return -EINVAL;
-   }
-
-   prtd->qmc_dai = qmc_dai;
-
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
hw_rule_channels_by_format = 
qmc_dai_hw_rule_capture_channels_by_format;
hw_rule_format_by_channels = 
qmc_dai_hw_rule_capture_format_by_channels;
@@ -468,7 +458,7 @@ static int qmc_dai_startup(struct snd_pcm_substream 
*substream,
  hw_rule_channels_by_format, qmc_dai,
  SNDRV_PCM_HW_PARAM_FORMAT, -1);
if (ret) {
-   dev_err(dai->dev, "Failed to add channels rule (%d)\n", ret);
+   dev_err(qmc_dai->dev, "Failed to add channels rule (%d)\n", 
ret);
return ret;
}
 
@@ -476,7 +466,7 @@ static int qmc_dai_startup(struct snd_pcm_substream 
*substream,
  hw_rule_format_by_channels, qmc_dai,
  SNDRV_PCM_HW_PARAM_CHANNELS, -1);
if (ret) {
-   dev_err(dai->dev, "Failed to add format rule (%d)\n", ret);
+   dev_err(qmc_dai->dev, "Failed to add format rule (%d)\n", ret);
return ret;
}
 
@@ -484,13 +474,30 @@ static int qmc_dai_startup(struct snd_pcm_substream 
*substream,
   SNDRV_PCM_HW_PARAM_FRAME_BITS,
   frame_bits);
if (ret < 0) {
-   dev_err(dai->dev, "Failed to add frame_bits constraint (%d)\n", 
ret);
+   dev_err(qmc_dai->dev, "Failed to add frame_bits constraint 
(%d)\n", ret);
return ret;
}
 
return 0;
 }
 
+static int qmc_dai_startup(struct snd_pcm_substream *substream,
+  struct snd_soc_dai *dai)
+{
+   struct qmc_dai_prtd *prtd = substream->runtime->private_data;
+   struct qmc_dai *qmc_dai;
+
+   qmc_dai = qmc_dai_get_data(dai);
+   if (!qmc_dai) {
+   dev_err(dai->dev, "Invalid dai\n");
+   return -EINVAL;
+   }
+
+   prtd->qmc_dai = qmc_dai;
+
+   return qmc_dai_constraints_interleaved(substream, qmc_dai);
+}
+
 static int qmc_dai_hw_params(struct snd_pcm_substream *substream,
 struct snd_pcm_hw_params *params,
 struct snd_soc_dai *dai)
-- 
2.45.0

[PATCH 08/10] soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles()

No function in the QMC API is available to get the number of phandles
present in a phandle list.

Fill this lack introducing qmc_chan_count_phandles().

Signed-off-by: Herve Codina 
---
 drivers/soc/fsl/qe/qmc.c | 13 +
 include/soc/fsl/qe/qmc.h |  2 ++
 2 files changed, 15 insertions(+)

diff --git a/drivers/soc/fsl/qe/qmc.c b/drivers/soc/fsl/qe/qmc.c
index e23d60018400..76bb496305a0 100644
--- a/drivers/soc/fsl/qe/qmc.c
+++ b/drivers/soc/fsl/qe/qmc.c
@@ -1777,6 +1777,19 @@ static struct qmc_chan *qmc_chan_get_from_qmc(struct 
device_node *qmc_np, unsign
return qmc_chan;
 }
 
+int qmc_chan_count_phandles(struct device_node *np, const char *phandles_name)
+{
+   int count;
+
+   /* phandles are fixed args phandles with one arg */
+   count = of_count_phandle_with_args(np, phandles_name, NULL);
+   if (count < 0)
+   return count;
+
+   return count / 2;
+}
+EXPORT_SYMBOL(qmc_chan_count_phandles);
+
 struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np,
   const char *phandles_name,
   int index)
diff --git a/include/soc/fsl/qe/qmc.h b/include/soc/fsl/qe/qmc.h
index 0fa7205145ce..294e42ea8d4c 100644
--- a/include/soc/fsl/qe/qmc.h
+++ b/include/soc/fsl/qe/qmc.h
@@ -16,6 +16,8 @@ struct device_node;
 struct device;
 struct qmc_chan;
 
+int qmc_chan_count_phandles(struct device_node *np, const char *phandles_name);
+
 struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np,
   const char *phandles_name,
   int index);
-- 
2.45.0

[PATCH 02/10] ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch

./scripts/checkpatch.pl --strict --codespell detected several issues
when running on the fsl_qmc_audio.c file:
  - CHECK: spaces preferred around that '*' (ctx:VxV)
  - CHECK: Alignment should match open parenthesis
  - CHECK: Comparison to NULL could be written "!prtd"
  - CHECK: spaces preferred around that '/' (ctx:VxV)
  - CHECK: Lines should not end with a '('
  - CHECK: Please don't use multiple blank lines
Some of them are present several times.

Fix all of these issues without any functional changes.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 65 +--
 1 file changed, 31 insertions(+), 34 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index dd90ef16fa97..917a32389f3d 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -54,7 +54,7 @@ static int qmc_audio_pcm_construct(struct snd_soc_component 
*component,
return ret;
 
snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, card->dev,
-  64*1024, 64*1024);
+  64 * 1024, 64 * 1024);
return 0;
 }
 
@@ -89,8 +89,8 @@ static void qmc_audio_pcm_write_complete(void *context)
prtd->period_ptr_submitted = prtd->dma_buffer_start;
 
ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-   qmc_audio_pcm_write_complete, prtd);
+   prtd->period_ptr_submitted, 
prtd->period_size,
+   qmc_audio_pcm_write_complete, prtd);
if (ret) {
dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
ret);
@@ -118,8 +118,8 @@ static void qmc_audio_pcm_read_complete(void *context, 
size_t length, unsigned i
prtd->period_ptr_submitted = prtd->dma_buffer_start;
 
ret = qmc_chan_read_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-   qmc_audio_pcm_read_complete, prtd);
+  prtd->period_ptr_submitted, 
prtd->period_size,
+  qmc_audio_pcm_read_complete, prtd);
if (ret) {
dev_err(prtd->qmc_dai->dev, "read_submit failed %d\n",
ret);
@@ -144,8 +144,8 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* Submit first chunk ... */
ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-   qmc_audio_pcm_write_complete, prtd);
+   prtd->period_ptr_submitted, 
prtd->period_size,
+   
qmc_audio_pcm_write_complete, prtd);
if (ret) {
dev_err(component->dev, "write_submit failed 
%d\n",
ret);
@@ -159,8 +159,8 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
 
/* ... and send it */
ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-   qmc_audio_pcm_write_complete, prtd);
+   prtd->period_ptr_submitted, 
prtd->period_size,
+   
qmc_audio_pcm_write_complete, prtd);
if (ret) {
dev_err(component->dev, "write_submit failed 
%d\n",
ret);
@@ -169,8 +169,8 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
} else {
/* Submit first chunk ... */
ret = qmc_chan_read_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-   qmc_audio_pcm_read_complete, prtd);
+  prtd->period_ptr_submitted, 
prtd->period_size,
+  qmc_audio_pcm_read_complete, 
prtd);
if (ret) {
dev_err(component->dev, "read_submit failed 
%d\n",
ret);
@@ -184,8 +184,8 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
 
/* ... and send it */
ret = qmc_chan_read_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, prtd->period_size,
-

[PATCH 00/10] Add support for non-interleaved mode in qmc_audio

The qmc_audio driver supports only audio in interleaved mode.
Non-interleaved mode can be easily supported using several QMC channel
per DAI. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.

In terms of constraints and settings, the interleaved and
non-interleaved modes are slightly different.

In interleaved mode:
  - The sample size should fit in the number of time-slots available for
the QMC channel.
  - The number of audio channels should fit in the number of time-slots
(taking into account the sample size) available for the QMC channel.

In non-interleaved mode:
  - The number of audio channels is the number of available QMC
channels.
  - Each QMC channel should have the same number of time-slots.
  - The sample size equals the number of time-slots of one QMC channel.

This series add support for the non-interleaved mode in the qmc_audio
driver and is composed of the following parts:
  - Patches 1 and 2: Fix some issues in the qmc_audio
  - Patches 3 to 6: Prepare qmc_audio for the non-interleaved mode
  - Patches 7 and 8: Extend the QMC driver API
  - Patches 9 and 10: The support for non-interleaved mode itself

Best regards,
Hervé

Herve Codina (10):
  ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned value
  ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch
  ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer
handling
  ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in
completion routines
  ASoC: fsl: fsl_qmc_audio: Introduce
qmc_audio_pcm_{read,write}_submit()
  ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved()
  soc: fsl: cpm1: qmc: Introduce functions to get a channel from a
phandle list
  soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles()
  dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC
channels per DAI
  ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode.

 .../bindings/sound/fsl,qmc-audio.yaml |  41 +-
 drivers/soc/fsl/qe/qmc.c  |  32 +-
 include/soc/fsl/qe/qmc.h  |  27 +-
 sound/soc/fsl/fsl_qmc_audio.c | 590 +-
 4 files changed, 505 insertions(+), 185 deletions(-)

-- 
2.45.0

[PATCH 10/10] ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode.

The current fsl_qmc_audio works in interleaved mode. The audio samples
are interleaved and all data are sent to (received from) one QMC
channel.

Using several QMC channels, non interleaved mode can be easily
supported. In that case, data related to ch0 are sent to (received from)
the first QMC channel, data related to ch1 use the next QMC channel and
so on up to the last channel.

In terms of constraints and settings, the two modes are slightly
different:
- Interleaved mode:
- The sample size should fit in the number of time-slots available
  for the QMC channel.
- The number of audio channels should fit in the number of
  time-slots (taking into account the sample size) available for the
  QMC channel.
- Non-interleaved mode:
- The number of audio channels is the number of available QMC
  channels.
- Each QMC channel should have the same number of time-slots.
- The sample size equals the number of time-slots of one QMC
  channel.

Add support for the non-interleaved mode allowing multiple QMC channel
per DAI. The DAI switches in non-interleaved mode when more that one QMC
channel is available.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 371 +++---
 1 file changed, 296 insertions(+), 75 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index f70c6c8eec4a..1560731c8372 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -29,7 +29,11 @@ struct qmc_dai {
struct device *dev;
unsigned int nb_tx_ts;
unsigned int nb_rx_ts;
-   struct qmc_dai_chan chan;
+
+   unsigned int nb_chans_avail;
+   unsigned int nb_chans_used_tx;
+   unsigned int nb_chans_used_rx;
+   struct qmc_dai_chan *chans;
 };
 
 struct qmc_audio {
@@ -50,7 +54,10 @@ struct qmc_dai_prtd {
dma_addr_t ch_dma_addr_current;
dma_addr_t ch_dma_addr_end;
size_t ch_dma_size;
+   size_t ch_dma_offset;
 
+   unsigned int channels;
+   DECLARE_BITMAP(chans_pending, 64);
struct snd_pcm_substream *substream;
 };
 
@@ -69,6 +76,17 @@ static int qmc_audio_pcm_construct(struct snd_soc_component 
*component,
return 0;
 }
 
+static bool qmc_audio_access_is_interleaved(snd_pcm_access_t access)
+{
+   switch (access) {
+   case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED:
+   case SNDRV_PCM_ACCESS_RW_INTERLEAVED:
+   return true;
+   default:
+   return false;
+   }
+};
+
 static int qmc_audio_pcm_hw_params(struct snd_soc_component *component,
   struct snd_pcm_substream *substream,
   struct snd_pcm_hw_params *params)
@@ -76,6 +94,14 @@ static int qmc_audio_pcm_hw_params(struct snd_soc_component 
*component,
struct snd_pcm_runtime *runtime = substream->runtime;
struct qmc_dai_prtd *prtd = substream->runtime->private_data;
 
+   /*
+* In interleaved mode, the driver uses one QMC channel for all audio
+* channels whereas in non-interleaved mode, it uses one QMC channel per
+* audio channel.
+*/
+   prtd->channels = qmc_audio_access_is_interleaved(params_access(params)) 
?
+   1 : params_channels(params);
+
prtd->substream = substream;
 
prtd->buffer_ended = 0;
@@ -83,9 +109,10 @@ static int qmc_audio_pcm_hw_params(struct snd_soc_component 
*component,
prtd->period_size = params_period_size(params);
 
prtd->ch_dma_addr_start = runtime->dma_addr;
-   prtd->ch_dma_addr_end = runtime->dma_addr + params_buffer_bytes(params);
+   prtd->ch_dma_offset = params_buffer_bytes(params) / prtd->channels;
+   prtd->ch_dma_addr_end = runtime->dma_addr + prtd->ch_dma_offset;
prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
-   prtd->ch_dma_size = params_period_bytes(params);
+   prtd->ch_dma_size = params_period_bytes(params) / prtd->channels;
 
return 0;
 }
@@ -94,16 +121,23 @@ static void qmc_audio_pcm_write_complete(void *context);
 
 static int qmc_audio_pcm_write_submit(struct qmc_dai_prtd *prtd)
 {
+   unsigned int i;
int ret;
 
-   ret = qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
-   prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
-   qmc_audio_pcm_write_complete,
-   &prtd->qmc_dai->chan);
-   if (ret) {
-   dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
-   ret);
-   return ret;
+   for (i = 0; i < prtd->channels; i++) {
+   bitmap_set(prtd->chans_pending, i, 1);
+
+   ret = qmc_chan_write_submit(prtd->qmc_dai->chans[i].qmc_chan,
+   prtd->ch_dma_addr_current + i * 
prtd->ch_dma_offset,
+   prtd->

[PATCH 01/10] ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned value

devm_kasprintf() can return a NULL pointer on failure but this returned
value is not checked.

Fix this lack and check the returned value.

Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio")
Cc: sta...@vger.kernel.org
Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index bfaaa451735b..dd90ef16fa97 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -604,6 +604,8 @@ static int qmc_audio_dai_parse(struct qmc_audio *qmc_audio, 
struct device_node *
 
qmc_dai->name = devm_kasprintf(qmc_audio->dev, GFP_KERNEL, "%s.%d",
   np->parent->name, qmc_dai->id);
+   if (!qmc_dai->name)
+   return -ENOMEM;
 
qmc_dai->qmc_chan = devm_qmc_chan_get_byphandle(qmc_audio->dev, np,
"fsl,qmc-chan");
-- 
2.45.0

[PATCH 04/10] ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion routines

The current QMC audio driver uses only one QMC channel per DAI. The
context used by QMC channel transfer (read and write) completion
routines does not contains any QMC channel and the only one available
per DAI is used to schedule the next transfer.
This works pretty well with only one QMC channel per DAI.

The future support for non-inlerleave mode will use several QMC channel
per DAI. In that case, QMC channel transfer completion routines need to
identify the QMC channel related to the completion.

In order to fill this lack, even if identifying the current QMC channel
among several QMC channels is not needed for the current code, add one
indirection level and introduce the qmc_dai_chan data structrure.
This structure contains the QMC channel involved in the completion and
refererences to the runtime context (capture and playback) used by the
DAI.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 72 +++
 1 file changed, 47 insertions(+), 25 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index e8281e548746..b07770257bad 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -17,13 +17,19 @@
 #include 
 #include 
 
+struct qmc_dai_chan {
+   struct qmc_dai_prtd *prtd_tx;
+   struct qmc_dai_prtd *prtd_rx;
+   struct qmc_chan *qmc_chan;
+};
+
 struct qmc_dai {
char *name;
int id;
struct device *dev;
-   struct qmc_chan *qmc_chan;
unsigned int nb_tx_ts;
unsigned int nb_rx_ts;
+   struct qmc_dai_chan chan;
 };
 
 struct qmc_audio {
@@ -86,9 +92,12 @@ static int qmc_audio_pcm_hw_params(struct snd_soc_component 
*component,
 
 static void qmc_audio_pcm_write_complete(void *context)
 {
-   struct qmc_dai_prtd *prtd = context;
+   struct qmc_dai_chan *chan = context;
+   struct qmc_dai_prtd *prtd;
int ret;
 
+   prtd = chan->prtd_tx;
+
prtd->buffer_ended += prtd->period_size;
if (prtd->buffer_ended >= prtd->buffer_size)
prtd->buffer_ended = 0;
@@ -97,9 +106,10 @@ static void qmc_audio_pcm_write_complete(void *context)
if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
-   ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
+   ret = qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
-   qmc_audio_pcm_write_complete, prtd);
+   qmc_audio_pcm_write_complete,
+   &prtd->qmc_dai->chan);
if (ret) {
dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
ret);
@@ -110,9 +120,12 @@ static void qmc_audio_pcm_write_complete(void *context)
 
 static void qmc_audio_pcm_read_complete(void *context, size_t length, unsigned 
int flags)
 {
-   struct qmc_dai_prtd *prtd = context;
+   struct qmc_dai_chan *chan = context;
+   struct qmc_dai_prtd *prtd;
int ret;
 
+   prtd = chan->prtd_rx;
+
if (length != prtd->ch_dma_size) {
dev_err(prtd->qmc_dai->dev, "read complete length = %zu, exp 
%zu\n",
length, prtd->ch_dma_size);
@@ -126,9 +139,10 @@ static void qmc_audio_pcm_read_complete(void *context, 
size_t length, unsigned i
if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
-   ret = qmc_chan_read_submit(prtd->qmc_dai->qmc_chan,
+   ret = qmc_chan_read_submit(prtd->qmc_dai->chan.qmc_chan,
   prtd->ch_dma_addr_current, prtd->ch_dma_size,
-  qmc_audio_pcm_read_complete, prtd);
+  qmc_audio_pcm_read_complete,
+  &prtd->qmc_dai->chan);
if (ret) {
dev_err(prtd->qmc_dai->dev, "read_submit failed %d\n",
ret);
@@ -151,10 +165,13 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+   prtd->qmc_dai->chan.prtd_tx = prtd;
+
/* Submit first chunk ... */
-   ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
+   ret = 
qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
-   
qmc_audio_pcm_write_complete, prtd);
+   
qmc_audio_pcm_write_complete,
+   &prtd->qmc_dai->chan);

[PATCH 09/10] dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAI

The QMC audio uses one QMC channel per DAI and uses this QMC channel to
transmit interleaved audio channel samples.

In order to work in non-interleave mode, a QMC audio DAI needs to use
multiple QMC channels. In that case, the DAI maps each QMC channel to
exactly one audio channel.

Allow QMC audio DAIs with multiple QMC channels attached.

Signed-off-by: Herve Codina 
---
 .../bindings/sound/fsl,qmc-audio.yaml | 41 ---
 1 file changed, 35 insertions(+), 6 deletions(-)

diff --git a/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml 
b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml
index b522ed7dcc51..a23e49198c37 100644
--- a/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml
@@ -12,7 +12,9 @@ maintainers:
 description: |
   The QMC audio is an ASoC component which uses QMC (QUICC Multichannel
   Controller) channels to transfer the audio data.
-  It provides as many DAI as the number of QMC channel used.
+  It provides several DAIs. For each DAI, the DAI is working in interleaved 
mode
+  if only one QMC channel is used by the DAI or it is working in 
non-interleaved
+  mode if several QMC channels are used by the DAI.
 
 allOf:
   - $ref: dai-common.yaml#
@@ -45,12 +47,19 @@ patternProperties:
   fsl,qmc-chan:
 $ref: /schemas/types.yaml#/definitions/phandle-array
 items:
-  - items:
-  - description: phandle to QMC node
-  - description: Channel number
+  items:
+- description: phandle to QMC node
+- description: Channel number
+minItems: 1
 description:
-  Should be a phandle/number pair. The phandle to QMC node and the QMC
-  channel to use for this DAI.
+  Should be a phandle/number pair list. The list of phandle to QMC node
+  and the QMC channel pair to use for this DAI.
+  If only one phandle/number pair is provided, this DAI works in
+  interleaved mode, i.e. audio channels for this DAI are interleaved in
+  the QMC channel. If more than one pair is provided, this DAI works
+  in non-interleave mode. In that case the first audio channel uses the
+  the first QMC channel, the second audio channel uses the second QMC
+  channel, etc...
 
 required:
   - reg
@@ -79,6 +88,11 @@ examples:
 reg = <17>;
 fsl,qmc-chan = <&qmc 17>;
 };
+dai@18 {
+reg = <18>;
+/* Non-interleaved mode */
+fsl,qmc-chan = <&qmc 18>, <&qmc 19>;
+};
 };
 
 sound {
@@ -115,4 +129,19 @@ examples:
 dai-tdm-slot-rx-mask = <0 0 1 0 1 0 1 0 1>;
 };
 };
+simple-audio-card,dai-link@2 {
+reg = <2>;
+format = "dsp_b";
+cpu {
+sound-dai = <&audio_controller 18>;
+};
+codec {
+sound-dai = <&codec3>;
+dai-tdm-slot-num = <2>;
+dai-tdm-slot-width = <8>;
+/* TS 9, 10 */
+dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0 0 1 1>;
+dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0 0 1 1>;
+};
+};
 };
-- 
2.45.0

[PATCH 05/10] ASoC: fsl: fsl_qmc_audio: Introduce qmc_audio_pcm_{read,write}_submit()

Submitting data to QMC channels is done in several places: transfer
completions and DAI start. The operation done is simple and consist in
one function call.

With the future introduction of support for non-interleaved mode,
submitting data will be more complex.

To avoid copy/paste of code in several places, introduce
qmc_audio_pcm_{read,write}_submit() whose goal is to handle this
data submission.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 93 +--
 1 file changed, 45 insertions(+), 48 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index b07770257bad..36145f1ddbf1 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -90,11 +90,29 @@ static int qmc_audio_pcm_hw_params(struct snd_soc_component 
*component,
return 0;
 }
 
+static void qmc_audio_pcm_write_complete(void *context);
+
+static int qmc_audio_pcm_write_submit(struct qmc_dai_prtd *prtd)
+{
+   int ret;
+
+   ret = qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
+   prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
+   qmc_audio_pcm_write_complete,
+   &prtd->qmc_dai->chan);
+   if (ret) {
+   dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
+   ret);
+   return ret;
+   }
+
+   return 0;
+}
+
 static void qmc_audio_pcm_write_complete(void *context)
 {
struct qmc_dai_chan *chan = context;
struct qmc_dai_prtd *prtd;
-   int ret;
 
prtd = chan->prtd_tx;
 
@@ -106,23 +124,33 @@ static void qmc_audio_pcm_write_complete(void *context)
if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
-   ret = qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
-   prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
-   qmc_audio_pcm_write_complete,
-   &prtd->qmc_dai->chan);
+   qmc_audio_pcm_write_submit(prtd);
+
+   snd_pcm_period_elapsed(prtd->substream);
+}
+
+static void qmc_audio_pcm_read_complete(void *context, size_t length, unsigned 
int flags);
+
+static int qmc_audio_pcm_read_submit(struct qmc_dai_prtd *prtd)
+{
+   int ret;
+
+   ret = qmc_chan_read_submit(prtd->qmc_dai->chan.qmc_chan,
+  prtd->ch_dma_addr_current, prtd->ch_dma_size,
+  qmc_audio_pcm_read_complete,
+  &prtd->qmc_dai->chan);
if (ret) {
-   dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
+   dev_err(prtd->qmc_dai->dev, "read_submit failed %d\n",
ret);
}
 
-   snd_pcm_period_elapsed(prtd->substream);
+   return 0;
 }
 
 static void qmc_audio_pcm_read_complete(void *context, size_t length, unsigned 
int flags)
 {
struct qmc_dai_chan *chan = context;
struct qmc_dai_prtd *prtd;
-   int ret;
 
prtd = chan->prtd_rx;
 
@@ -139,14 +167,7 @@ static void qmc_audio_pcm_read_complete(void *context, 
size_t length, unsigned i
if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
-   ret = qmc_chan_read_submit(prtd->qmc_dai->chan.qmc_chan,
-  prtd->ch_dma_addr_current, prtd->ch_dma_size,
-  qmc_audio_pcm_read_complete,
-  &prtd->qmc_dai->chan);
-   if (ret) {
-   dev_err(prtd->qmc_dai->dev, "read_submit failed %d\n",
-   ret);
-   }
+   qmc_audio_pcm_read_submit(prtd);
 
snd_pcm_period_elapsed(prtd->substream);
 }
@@ -168,15 +189,9 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component 
*component,
prtd->qmc_dai->chan.prtd_tx = prtd;
 
/* Submit first chunk ... */
-   ret = 
qmc_chan_write_submit(prtd->qmc_dai->chan.qmc_chan,
-   prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
-   
qmc_audio_pcm_write_complete,
-   &prtd->qmc_dai->chan);
-   if (ret) {
-   dev_err(component->dev, "write_submit failed 
%d\n",
-   ret);
+   ret = qmc_audio_pcm_write_submit(prtd);
+   if (ret)
return ret;
-   }
 
/* ... prepare next one ... */
prtd->ch_dma_addr_current += prtd->ch_dma_size;
@@ -184,28 +199,16 @@ static int qmc_audio_pcm_trigger(struct

[PATCH 03/10] ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handling

The driver mixes some internal values for channel DMA buffer handling
and PCM pointer handling. In the currently supported interleaved mode,
this mix does not lead to any issues but in order to prepare the
support for the non-interleaved mode, having them clearly separated will
ease the support and avoid additional computation to convert values used
in channel DMA buffer management in values usable for PCM pointer.

Use a specific set of variable for PCM pointer handling and an other set
for channel DMA buffer.

Signed-off-by: Herve Codina 
---
 sound/soc/fsl/fsl_qmc_audio.c | 84 +++
 1 file changed, 46 insertions(+), 38 deletions(-)

diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c
index 917a32389f3d..e8281e548746 100644
--- a/sound/soc/fsl/fsl_qmc_audio.c
+++ b/sound/soc/fsl/fsl_qmc_audio.c
@@ -35,11 +35,16 @@ struct qmc_audio {
 
 struct qmc_dai_prtd {
struct qmc_dai *qmc_dai;
-   dma_addr_t dma_buffer_start;
-   dma_addr_t period_ptr_submitted;
-   dma_addr_t period_ptr_ended;
-   dma_addr_t dma_buffer_end;
-   size_t period_size;
+
+   snd_pcm_uframes_t buffer_ended;
+   snd_pcm_uframes_t buffer_size;
+   snd_pcm_uframes_t period_size;
+
+   dma_addr_t ch_dma_addr_start;
+   dma_addr_t ch_dma_addr_current;
+   dma_addr_t ch_dma_addr_end;
+   size_t ch_dma_size;
+
struct snd_pcm_substream *substream;
 };
 
@@ -65,13 +70,17 @@ static int qmc_audio_pcm_hw_params(struct snd_soc_component 
*component,
struct snd_pcm_runtime *runtime = substream->runtime;
struct qmc_dai_prtd *prtd = substream->runtime->private_data;
 
-   prtd->dma_buffer_start = runtime->dma_addr;
-   prtd->dma_buffer_end = runtime->dma_addr + params_buffer_bytes(params);
-   prtd->period_size = params_period_bytes(params);
-   prtd->period_ptr_submitted = prtd->dma_buffer_start;
-   prtd->period_ptr_ended = prtd->dma_buffer_start;
prtd->substream = substream;
 
+   prtd->buffer_ended = 0;
+   prtd->buffer_size = params_buffer_size(params);
+   prtd->period_size = params_period_size(params);
+
+   prtd->ch_dma_addr_start = runtime->dma_addr;
+   prtd->ch_dma_addr_end = runtime->dma_addr + params_buffer_bytes(params);
+   prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
+   prtd->ch_dma_size = params_period_bytes(params);
+
return 0;
 }
 
@@ -80,16 +89,16 @@ static void qmc_audio_pcm_write_complete(void *context)
struct qmc_dai_prtd *prtd = context;
int ret;
 
-   prtd->period_ptr_ended += prtd->period_size;
-   if (prtd->period_ptr_ended >= prtd->dma_buffer_end)
-   prtd->period_ptr_ended = prtd->dma_buffer_start;
+   prtd->buffer_ended += prtd->period_size;
+   if (prtd->buffer_ended >= prtd->buffer_size)
+   prtd->buffer_ended = 0;
 
-   prtd->period_ptr_submitted += prtd->period_size;
-   if (prtd->period_ptr_submitted >= prtd->dma_buffer_end)
-   prtd->period_ptr_submitted = prtd->dma_buffer_start;
+   prtd->ch_dma_addr_current += prtd->ch_dma_size;
+   if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
+   prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
ret = qmc_chan_write_submit(prtd->qmc_dai->qmc_chan,
-   prtd->period_ptr_submitted, 
prtd->period_size,
+   prtd->ch_dma_addr_current, 
prtd->ch_dma_size,
qmc_audio_pcm_write_complete, prtd);
if (ret) {
dev_err(prtd->qmc_dai->dev, "write_submit failed %d\n",
@@ -104,21 +113,21 @@ static void qmc_audio_pcm_read_complete(void *context, 
size_t length, unsigned i
struct qmc_dai_prtd *prtd = context;
int ret;
 
-   if (length != prtd->period_size) {
+   if (length != prtd->ch_dma_size) {
dev_err(prtd->qmc_dai->dev, "read complete length = %zu, exp 
%zu\n",
-   length, prtd->period_size);
+   length, prtd->ch_dma_size);
}
 
-   prtd->period_ptr_ended += prtd->period_size;
-   if (prtd->period_ptr_ended >= prtd->dma_buffer_end)
-   prtd->period_ptr_ended = prtd->dma_buffer_start;
+   prtd->buffer_ended += prtd->period_size;
+   if (prtd->buffer_ended >= prtd->buffer_size)
+   prtd->buffer_ended = 0;
 
-   prtd->period_ptr_submitted += prtd->period_size;
-   if (prtd->period_ptr_submitted >= prtd->dma_buffer_end)
-   prtd->period_ptr_submitted = prtd->dma_buffer_start;
+   prtd->ch_dma_addr_current += prtd->ch_dma_size;
+   if (prtd->ch_dma_addr_current >= prtd->ch_dma_addr_end)
+   prtd->ch_dma_addr_current = prtd->ch_dma_addr_start;
 
ret = qmc_chan_read_submit(prtd->qmc_dai->qmc_chan,
-  prtd->period_ptr_submitted, 
prtd->period_size,
+

Re: [PATCH] KVM: PPC: Book3S HV: Prevent UAF in kvm_spapr_tce_attach_iommu_group()

On Fri, 14 Jun 2024 22:29:10 +1000, Michael Ellerman wrote:
> Al reported a possible use-after-free (UAF) in 
> kvm_spapr_tce_attach_iommu_group().
> 
> It looks up `stt` from tablefd, but then continues to use it after doing
> fdput() on the returned fd. After the fdput() the tablefd is free to be
> closed by another thread. The close calls kvm_spapr_tce_release() and
> then release_spapr_tce_table() (via call_rcu()) which frees `stt`.
> 
> [...]

Applied to powerpc/fixes.

[1/1] KVM: PPC: Book3S HV: Prevent UAF in kvm_spapr_tce_attach_iommu_group()
  https://git.kernel.org/powerpc/c/a986fa57fd81a1430e00b3c6cf8a325d6f894a63

cheers

Re: [PATCH] powerpc/crypto: Add generated P8 asm to .gitignore

On Mon, 03 Jun 2024 08:01:03 -0500, Nathan Lynch wrote:
> Looks like drivers/crypto/vmx/.gitignore should have been merged into
> arch/powerpc/crypto/.gitignore as part of commit
> 109303336a0c ("crypto: vmx - Move to arch/powerpc/crypto") so that all
> generated asm files are ignored.
> 
> 

Applied to powerpc/fixes.

[1/1] powerpc/crypto: Add generated P8 asm to .gitignore
  https://git.kernel.org/powerpc/c/2b85b7fb1376481f7d4c2cf92e5da942f06b2547

cheers

Re: (subset) [PATCH 0/6] defconfig: drop RT_GROUP_SCHED=y

On Thu, 30 May 2024 19:19:48 +0800, Celeste Liu wrote:
> For cgroup v1, if turned on, and there's any cgroup in the "cpu" hierarchy it
> needs an RT budget assigned, otherwise the processes in it will not be able to
> get RT at all. The problem with RT group scheduling is that it requires the
> budget assigned but there's no way we could assign a default budget, since the
> values to assign are both upper and lower time limits, are absolute, and need 
> to
> be sum up to < 1 for each individal cgroup. That means we cannot really come 
> up
> with values that would work by default in the general case.[1]
> 
> [...]

Patch 4 applied to powerpc/next.

[4/6] powerpc: defconfig: drop RT_GROUP_SCHED=y from ppc6xx_defconfig
  https://git.kernel.org/powerpc/c/2bac6caee94e25f59ee47e2d365d7e07465089ba

cheers

Re: [PATCH v2] arch/powerpc: Remove unused cede related functions

On Tue, 14 May 2024 18:54:55 +0530, Gautam Menghani wrote:
> Remove extended_cede_processor() and its helpers as
> extended_cede_processor() has no callers since
> commit 48f6e7f6d948("powerpc/pseries: remove cede offline state for CPUs")
> 
> 

Applied to powerpc/next.

[1/1] arch/powerpc: Remove unused cede related functions
  https://git.kernel.org/powerpc/c/214f33fcf656bf1be3f9f03d58fda067cdf7eecc

cheers

Re: [PATCH] powerpc/pseries/iommu: Split Dynamic DMA Window to be used in Hybrid mode

On Mon, 13 May 2024 20:46:08 -0500, Gaurav Batra wrote:
> Dynamic DMA Window (DDW) supports TCEs that are backed by 2MB page size.
> In most configurations, DDW is big enough to pre-map all of LPAR memory
> for IO. Pre-mapping of memory for DMA results in improvements in IO
> performance.
> 
> Persistent memory, vPMEM, can be assigned to an LPAR as well. vPMEM is not
> contiguous with LPAR memory and usually is assigned at high memory
> addresses.  This makes is not possible to pre-map both vPMEM and LPAR
> memory in the same DDW.
> 
> [...]

Applied to powerpc/next.

[1/1] powerpc/pseries/iommu: Split Dynamic DMA Window to be used in Hybrid mode
  https://git.kernel.org/powerpc/c/ff5163bb7000a0254ffdd7b50cb6df43add94f33

cheers

Re: [PATCH v6] powerpc/pseries/vas: Use usleep_range() to support HCALL delay

On Mon, 15 Jan 2024 21:59:10 -0800, Haren Myneni wrote:
> VAS allocate, modify and deallocate HCALLs returns
> H_LONG_BUSY_ORDER_1_MSEC or H_LONG_BUSY_ORDER_10_MSEC for busy
> delay and expects OS to reissue HCALL after that delay. But using
> msleep() will often sleep at least 20 msecs even though the
> hypervisor suggests OS reissue these HCALLs after 1 or 10msecs.
> 
> The open and close VAS window functions hold mutex and then issue
> these HCALLs. So these operations can take longer than the
> necessary when multiple threads issue open or close window APIs
> simultaneously, especially might affect the performance in the
> case of repeat open/close APIs for each compression request.
> 
> [...]

Applied to powerpc/next.

[1/1] powerpc/pseries/vas: Use usleep_range() to support HCALL delay
  https://git.kernel.org/powerpc/c/43ac9f5cd457bb01930f87448ddaaae455f8a8cf

cheers

Re: [PATCH] powerpc/mm/drmem: Silence drmem_init() early return

On Mon, 03 Jun 2024 14:31:32 -0500, Nathan Lynch wrote:
> It's not an error or noteworthy condition if the
> "ibm,dynamic-reconfiguration-memory" node isn't present.
> 
> Drop the needless message.
> 
> 

Applied to powerpc/next.

[1/1] powerpc/mm/drmem: Silence drmem_init() early return
  https://git.kernel.org/powerpc/c/11e6e6d8bf8f908468bac0447727e3f3923c8512

cheers

Re: [PATCH v2 0/1] powerpc/numa: Make cpu/memory less numa-node online

On Fri, 17 May 2024 19:55:21 +0530, Nilay Shroff wrote:
> On NUMA aware system, we make a numa-node online only if that node is
> attached to cpu/memory. However it's possible that we have some PCI/IO
> device affinitized to a numa-node which is not currently online. In such
> case we set the numa-node id of the corresponding PCI device to -1
> (NUMA_NO_NODE). Not assigning the correct numa-node id to PCI device may
> impact the performance of such device. For instance, we have a multi
> controller NVMe disk where each controller of the disk is attached to
> different PHB (PCI host bridge). Each of these PHBs has numa-node id
> assigned during PCI enumeration. During PCI enumeration if we find that
> the numa-node is not online then we set the numa-node id of the PHB to -1.
> If we create shared namespace and attach to multi controller NVMe disk
> then that namespace could be accessed through each controller and as each
> controller is connected to different PHBs, it's possible to access the
> same namespace using multiple PCI channel. While sending IO to a shared
> namespace, NVMe driver would calculate the optimal IO path using numa-node
> distance. However if the numa-node id is not correctly assigned to NVMe
> PCIe controller then it's possible that driver would calculate incorrect
> NUMA distance and hence select the non-optimal path for sending IO. If
> this happens then we could potentially observe the degraded IO performance.
> 
> [...]

Applied to powerpc/next.

[1/1] powerpc/numa: Online a node if PHB is attached.
  https://git.kernel.org/powerpc/c/11981816e3614156a1fe14a1e8e77094ea46c7d5

cheers

[PATCHv5 0/9] ASoC: fsl-asoc-card: add S/PDIF controller

Hello,

This is the v5 of the series of patches aiming to make the machine
driver `fsl-asoc-card` compatible with S/PDIF controllers on imx boards.
The main goal is to allow the use of S/PDIF controllers with ASRC
modules.

The `imx-spdif` machine driver already has specific support for S/PDIF
controllers but doesn't support using an ASRC with it. However, the
`fsl-asoc-card` machine driver has the necessary code to create a sound
card which can use an ASRC module.
It is then possible to extend the support for S/PDIF audio cards by
merging the `imx-spdif` driver into `fsl-asoc-card`.

The first patch fixes a NULL pointer dereference in `fsl-asoc-card`
which occurs when the function "fsl_asoc_card_audmux_init()" prints
error logs.

The next three patches adapt the `fsl-asoc-card` driver to support
multiple codec use cases.
The driver can get 2 codec phandles from the device tree, and
codec-related variables are doubled.
`for_each_codecs` macros are also used when possible to ease adding
other multi-codec use cases in the future.
This makes possible to use the two S/PDIF dummy codec drivers
`spdif_receiver` and `spdif_transmitter` instead of `snd-soc-dummy`,
which was used in `imx-spdif`.

The fifth patch merges the S/PDIF support from `imx-spdif` to
`fsl-asoc-card`.
`fsl-asoc-card` offers the same functionalities as `imx-spdif` did, but
this merge also extends the S/PDIF support with the possibility of using
an ASRC.
Compatible "fsl,imx-audio-spdif" is kept, but some DT properties have to
be updated in device trees to match those of `fsl-asoc-card`:
* "spdif-controller" is now "audio-cpu" in `fsl-asoc-card`.
* "spdif-in" and "spdif-out" are now replaced by the "audio-codec"
property of `fsl-asoc-card`. S/PDIF specific dummy codecs should be
used with `fsl-asoc-card`, and therefore declared in device trees.

The last four patches update the device tree bindings of
`fsl-asoc-card`, remove the now obsolete binding of `imx-spdif`, and
update all device trees using the `fsl,imx-audio-spdif` compatible.

This series of patches was successfully built for arm64 and x86 on top
of the latest??"for-next" branch of the ASoC git tree on the 19th of June
2024.
These modifications have also been tested on an i.MX8MN evaluation board
with a linux kernel RT v6.1.26-rt8.

If you have any questions or remarks about these commits, don't hesitate
to reply to this message.

Best regards,
Elinor Montmasson

Changelog:
v4 -> v5:
* Focus the contribution to bringing S/PDIF / ASRC support.
* Instead of creating a new compatible for the S/PDIF `fsl-asoc-card`
support, merge the driver `imx-spdif` into `fsl-asoc-card`, and keep
the compatible. It preserves the base S/PDIF audio card support but
also extends it with the possibility to use an ASRC.
It also reduces code and driver duplication.
* Following driver merge, adapt device trees using "fsl,imx-audio-spdif"
compatible.
* Use more `for_each_codecs` macros in `fsl-asoc-card` when adding
multi-codec support.
* Remove patches about new device-tree bindings that were not relevant
for an S/PDIF specific support.
* Improve DT schema changes.
* Move `priv->pdev` assignment earlier in "fsl_asoc_card_probe()" to fix
a NULL pointer dereference in "fsl_asoc_card_audmux_init()".
* v4 patch series at :
https://lore.kernel.org/all/20240515135411.34-1-elinor.montmas...@savoirfairelinux.com/

v3 -> v4:
* Use the standard TDM bidings, as defined in "tdm-slot.txt", for the
new optional DT bindings setting the TDM slots number and width.
* Use the clock DT bindings to optionally specify the CPU DAI system
clock frequency, instead of a dedicated new binding.
* Rename the new DT binding "cpu-sysclk-dir-out" to
"cpu-system-clock-direction-out" to better follow the style of the
simple-card driver.
* Merge TX an RX bindings for CPU DAI system-clock, to better follow the
style of the simple-card driver, and also as there was no use case in
fsl-asoc-card where TX and RX settings had to be different.
* Add the documentation for the new bindings in the new DT schema
bindings documentation. Also add an example with the generic codec.
* v3 patch series at :
https://lore.kernel.org/alsa-devel/20231218124058.2047167-1-elinor.montmas...@savoirfairelinux.com/

v2 -> v3:
* When the bitmaster or framemaster are retrieved from the device tree,
the driver will now compare them with the two codecs possibly given in
device tree, and not just the first codec.
* Improve driver modifications to use multiple codecs for better
integration of future multi-codec use cases:
* Use `for_each_codec` macros when possible.
* `fsl_asoc_card_priv` struct now has 2 `codec_priv` as the driver
can currently retrieve 2 codec phandles from the device tree.
* Fix subject of patch 10/10 to follow the style of the subsystem
* v2 patch series at:
https://lore.kernel.org/alsa-devel/20231027144734.3654829-1-elinor.montmas...@savoirfairelinux.com/

v1 -> v2:
* Replace use of the dummy cod

[PATCHv5 1/9] ASoC: fsl-asoc-card: set priv->pdev before using it

priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.

fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.

Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with 
ASRC support")
Signed-off-by: Elinor Montmasson 
---
 sound/soc/fsl/fsl-asoc-card.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5ddc0c2fe53f..eb67689dcd6e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -559,6 +559,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
if (!priv)
return -ENOMEM;
 
+   priv->pdev = pdev;
+
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
/* Give a chance to old DT binding */
if (!cpu_np)
@@ -787,7 +789,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
 
/* Initialize sound card */
-   priv->pdev = pdev;
priv->card.dev = &pdev->dev;
priv->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&priv->card, "model");
-- 
2.34.1

[PATCHv5 2/9] ASoC: fsl-asoc-card: add support for dai links with multiple codecs

Add support for dai links using multiple codecs for multi-codec
use cases.

Co-developed-by: Philip-Dylan Gleonec 

Signed-off-by: Philip-Dylan Gleonec 
Signed-off-by: Elinor Montmasson 
---
 sound/soc/fsl/fsl-asoc-card.c | 19 +++
 1 file changed, 11 insertions(+), 8 deletions(-)

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index eb67689dcd6e..e8003fbc8092 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -542,6 +542,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card 
*card)
 static int fsl_asoc_card_probe(struct platform_device *pdev)
 {
struct device_node *cpu_np, *codec_np, *asrc_np;
+   struct snd_soc_dai_link_component *codec_comp;
struct device_node *np = pdev->dev.of_node;
struct platform_device *asrc_pdev = NULL;
struct device_node *bitclkprovider = NULL;
@@ -552,6 +553,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
const char *codec_dai_name;
const char *codec_dev_name;
u32 asrc_fmt = 0;
+   int codec_idx;
u32 width;
int ret;
 
@@ -816,10 +818,10 @@ static int fsl_asoc_card_probe(struct platform_device 
*pdev)
 
/* Normal DAI Link */
priv->dai_link[0].cpus->of_node = cpu_np;
-   priv->dai_link[0].codecs->dai_name = codec_dai_name;
+   priv->dai_link[0].codecs[0].dai_name = codec_dai_name;
 
if (!fsl_asoc_card_is_ac97(priv))
-   priv->dai_link[0].codecs->of_node = codec_np;
+   priv->dai_link[0].codecs[0].of_node = codec_np;
else {
u32 idx;
 
@@ -830,11 +832,11 @@ static int fsl_asoc_card_probe(struct platform_device 
*pdev)
goto asrc_fail;
}
 
-   priv->dai_link[0].codecs->name =
+   priv->dai_link[0].codecs[0].name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
   "ac97-codec.%u",
   (unsigned int)idx);
-   if (!priv->dai_link[0].codecs->name) {
+   if (!priv->dai_link[0].codecs[0].name) {
ret = -ENOMEM;
goto asrc_fail;
}
@@ -848,10 +850,11 @@ static int fsl_asoc_card_probe(struct platform_device 
*pdev)
/* DPCM DAI Links only if ASRC exists */
priv->dai_link[1].cpus->of_node = asrc_np;
priv->dai_link[1].platforms->of_node = asrc_np;
-   priv->dai_link[2].codecs->dai_name = codec_dai_name;
-   priv->dai_link[2].codecs->of_node = codec_np;
-   priv->dai_link[2].codecs->name =
-   priv->dai_link[0].codecs->name;
+   for_each_link_codecs((&(priv->dai_link[2])), codec_idx, 
codec_comp) {
+   codec_comp->dai_name = 
priv->dai_link[0].codecs[codec_idx].dai_name;
+   codec_comp->of_node = 
priv->dai_link[0].codecs[codec_idx].of_node;
+   codec_comp->name = 
priv->dai_link[0].codecs[codec_idx].name;
+   }
priv->dai_link[2].cpus->of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->card.num_links = 3;
-- 
2.34.1

[PATCHv5 3/9] ASoC: fsl-asoc-card: add second dai link component for codecs

Add a second dai link component for codecs that will be used for use
cases with 2 codecs.
It is needed for future integration of the SPDIF support, which will
use spdif_receiver and spdif_transmitter drivers.

To prevent deferring in use cases using only one codec, also set
by default the number of codecs to 1 for the relevant dai links.

Co-developed-by: Philip-Dylan Gleonec 

Signed-off-by: Philip-Dylan Gleonec 
Signed-off-by: Elinor Montmasson 
---
 sound/soc/fsl/fsl-asoc-card.c | 6 --
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index e8003fbc8092..805e2030bde4 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -296,7 +296,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime 
*rtd,
 
 SND_SOC_DAILINK_DEFS(hifi,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
-   DAILINK_COMP_ARRAY(COMP_EMPTY()),
+   DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
 SND_SOC_DAILINK_DEFS(hifi_fe,
@@ -306,7 +306,7 @@ SND_SOC_DAILINK_DEFS(hifi_fe,
 
 SND_SOC_DAILINK_DEFS(hifi_be,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
-   DAILINK_COMP_ARRAY(COMP_EMPTY()));
+   DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()));
 
 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
/* Default ASoC DAI Link*/
@@ -622,6 +622,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
memcpy(priv->dai_link, fsl_asoc_card_dai,
   sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+   priv->dai_link[0].num_codecs = 1;
+   priv->dai_link[2].num_codecs = 1;
 
priv->card.dapm_routes = audio_map;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
-- 
2.34.1

[PATCHv5 4/9] ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links

Adapt the driver to work with configurations using two codecs or more.
Modify fsl_asoc_card_probe() to handle use cases where 2 codecs are
given in the device tree.
This will be needed to add support for the SPDIF.

Use cases using one codec will ignore any given codecs other than the
first.

Co-developed-by: Philip-Dylan Gleonec 

Signed-off-by: Philip-Dylan Gleonec 
Signed-off-by: Elinor Montmasson 
---
 sound/soc/fsl/fsl-asoc-card.c | 279 --
 1 file changed, 161 insertions(+), 118 deletions(-)

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 805e2030bde4..87329731e02d 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -99,7 +99,7 @@ struct fsl_asoc_card_priv {
struct simple_util_jack hp_jack;
struct simple_util_jack mic_jack;
struct platform_device *pdev;
-   struct codec_priv codec_priv;
+   struct codec_priv codec_priv[2];
struct cpu_priv cpu_priv;
struct snd_soc_card card;
u8 streams;
@@ -172,10 +172,12 @@ static int fsl_asoc_card_hw_params(struct 
snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-   struct codec_priv *codec_priv = &priv->codec_priv;
+   struct codec_priv *codec_priv;
+   struct snd_soc_dai *codec_dai;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
struct device *dev = rtd->card->dev;
unsigned int pll_out;
+   int codec_idx;
int ret;
 
priv->sample_rate = params_rate(params);
@@ -208,28 +210,32 @@ static int fsl_asoc_card_hw_params(struct 
snd_pcm_substream *substream,
}
 
/* Specific configuration for PLL */
-   if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
-   if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
-   pll_out = priv->sample_rate * 384;
-   else
-   pll_out = priv->sample_rate * 256;
+   for_each_rtd_codec_dais(rtd, codec_idx, codec_dai) {
+   codec_priv = &priv->codec_priv[codec_idx];
 
-   ret = snd_soc_dai_set_pll(snd_soc_rtd_to_codec(rtd, 0),
- codec_priv->pll_id,
- codec_priv->mclk_id,
- codec_priv->mclk_freq, pll_out);
-   if (ret) {
-   dev_err(dev, "failed to start FLL: %d\n", ret);
-   goto fail;
-   }
+   if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
+   if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+   pll_out = priv->sample_rate * 384;
+   else
+   pll_out = priv->sample_rate * 256;
 
-   ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0),
-codec_priv->fll_id,
-pll_out, SND_SOC_CLOCK_IN);
+   ret = snd_soc_dai_set_pll(codec_dai,
+   codec_priv->pll_id,
+   codec_priv->mclk_id,
+   codec_priv->mclk_freq, pll_out);
+   if (ret) {
+   dev_err(dev, "failed to start FLL: %d\n", ret);
+   goto fail;
+   }
 
-   if (ret && ret != -ENOTSUPP) {
-   dev_err(dev, "failed to set SYSCLK: %d\n", ret);
-   goto fail;
+   ret = snd_soc_dai_set_sysclk(codec_dai,
+   codec_priv->fll_id,
+   pll_out, SND_SOC_CLOCK_IN);
+
+   if (ret && ret != -ENOTSUPP) {
+   dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+   goto fail;
+   }
}
}
 
@@ -244,28 +250,34 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream 
*substream)
 {
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
-   struct codec_priv *codec_priv = &priv->codec_priv;
+   struct codec_priv *codec_priv;
+   struct snd_soc_dai *codec_dai;
struct device *dev = rtd->card->dev;
+   int codec_idx;
int ret;
 
priv->streams &= ~BIT(substream->stream);
 
-   if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 
0) {
-   /* Force freq to be free_freq to avoid error message in codec */
-

[PATCHv5 5/9] ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c

The imx-spdif machine driver creates audio card to directly use an
S/PDIF device. However, it doesn't support interacting with an ASRC.
fsl-asoc-card already has the support to create audio card which can
use the ASRC.

Merge the S/PDIF support from imx-spdif into fsl-asoc-card to extend
the support of S/PDIF audio card with the use of ASRC devices.
It also substitutes the use of the dummy codec in imx-spdif with the
existing spdif_transmitter and spdif_receiver codec drivers.

Signed-off-by: Elinor Montmasson 
---
 arch/arm/configs/imx_v6_v7_defconfig |   1 -
 arch/arm64/configs/defconfig |   1 -
 sound/soc/fsl/Kconfig|  10 +--
 sound/soc/fsl/Makefile   |   2 -
 sound/soc/fsl/fsl-asoc-card.c|  58 +++
 sound/soc/fsl/imx-spdif.c| 103 ---
 6 files changed, 59 insertions(+), 116 deletions(-)
 delete mode 100644 sound/soc/fsl/imx-spdif.c

diff --git a/arch/arm/configs/imx_v6_v7_defconfig 
b/arch/arm/configs/imx_v6_v7_defconfig
index cf2480dce285..ac5ae621b2af 100644
--- a/arch/arm/configs/imx_v6_v7_defconfig
+++ b/arch/arm/configs/imx_v6_v7_defconfig
@@ -311,7 +311,6 @@ CONFIG_SND_IMX_SOC=y
 CONFIG_SND_SOC_EUKREA_TLV320=y
 CONFIG_SND_SOC_IMX_ES8328=y
 CONFIG_SND_SOC_IMX_SGTL5000=y
-CONFIG_SND_SOC_IMX_SPDIF=y
 CONFIG_SND_SOC_FSL_ASOC_CARD=y
 CONFIG_SND_SOC_AC97_CODEC=y
 CONFIG_SND_SOC_CS42XX8_I2C=y
diff --git a/arch/arm64/configs/defconfig b/arch/arm64/configs/defconfig
index 57a9abe78ee4..a6c9688fee0e 100644
--- a/arch/arm64/configs/defconfig
+++ b/arch/arm64/configs/defconfig
@@ -940,7 +940,6 @@ CONFIG_SND_SOC_FSL_MICFIL=m
 CONFIG_SND_SOC_FSL_EASRC=m
 CONFIG_SND_IMX_SOC=m
 CONFIG_SND_SOC_IMX_SGTL5000=m
-CONFIG_SND_SOC_IMX_SPDIF=m
 CONFIG_SND_SOC_FSL_ASOC_CARD=m
 CONFIG_SND_SOC_IMX_AUDMIX=m
 CONFIG_SND_SOC_MT8183=m
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9a371d4496c2..e3b2bfb016c8 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -296,15 +296,6 @@ config SND_SOC_IMX_SGTL5000
  SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer
  driver.
 
-config SND_SOC_IMX_SPDIF
-   tristate "SoC Audio support for i.MX boards with S/PDIF"
-   select SND_SOC_IMX_PCM_DMA
-   select SND_SOC_FSL_SPDIF
-   help
- SoC Audio support for i.MX boards with S/PDIF
- Say Y if you want to add support for SoC audio on an i.MX board with
- a S/DPDIF.
-
 config SND_SOC_FSL_ASOC_CARD
tristate "Generic ASoC Sound Card with ASRC support"
depends on OF && I2C
@@ -316,6 +307,7 @@ config SND_SOC_FSL_ASOC_CARD
select SND_SOC_FSL_ESAI
select SND_SOC_FSL_SAI
select SND_SOC_FSL_SSI
+   select SND_SOC_FSL_SPDIF
select SND_SOC_TLV320AIC31XX
select SND_SOC_WM8994
select MFD_WM8994
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 2fe78eed3a48..1ae181b24a88 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -65,7 +65,6 @@ obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o
 snd-soc-eukrea-tlv320-y := eukrea-tlv320.o
 snd-soc-imx-es8328-y := imx-es8328.o
 snd-soc-imx-sgtl5000-y := imx-sgtl5000.o
-snd-soc-imx-spdif-y := imx-spdif.o
 snd-soc-imx-audmix-y := imx-audmix.o
 snd-soc-imx-hdmi-y := imx-hdmi.o
 snd-soc-imx-rpmsg-y := imx-rpmsg.o
@@ -74,7 +73,6 @@ snd-soc-imx-card-y := imx-card.o
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
-obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o
 obj-$(CONFIG_SND_SOC_IMX_HDMI) += snd-soc-imx-hdmi.o
 obj-$(CONFIG_SND_SOC_IMX_RPMSG) += snd-soc-imx-rpmsg.o
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 87329731e02d..c0e600525680 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -477,6 +477,59 @@ static int fsl_asoc_card_audmux_init(struct device_node 
*np,
return 0;
 }
 
+static int fsl_asoc_card_spdif_init(struct device_node *codec_np[],
+   struct device_node *cpu_np,
+   const char *codec_dai_name[],
+   struct fsl_asoc_card_priv *priv)
+{
+   struct device *dev = &priv->pdev->dev;
+
+   if (!of_node_name_eq(cpu_np, "spdif")) {
+   dev_err(dev, "CPU phandle invalid, should be an SPDIF 
device\n");
+   return -EINVAL;
+   }
+
+   priv->dai_link[0].playback_only = true;
+   priv->dai_link[0].capture_only = true;
+
+   for (int i = 0; i < 2; i++) {
+   if (!codec_np[i])
+   break;
+
+   if (of_device_is_compatible(codec_np[i], "linux,spdif-dit")) {
+   priv->dai_link[0].capture_only = false;
+   codec_dai_name[i] = "dit-hifi";
+

[PATCHv5 6/9] ASoC: dt-bindings: fsl-asoc-card: add compatible string for spdif

The S/PDIF audio card support was merged from imx-spdif into the
fsl-asoc-card driver, making it possible to use an S/PDIF with an ASRC.
Add the new compatible and update properties.

Signed-off-by: Elinor Montmasson 
---
 .../bindings/sound/fsl-asoc-card.yaml | 30 ---
 1 file changed, 26 insertions(+), 4 deletions(-)

diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml 
b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
index 9922664d5ccc..f2e28b32808e 100644
--- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml
@@ -33,6 +33,7 @@ properties:
   - items:
   - enum:
   - fsl,imx-sgtl5000
+  - fsl,imx-sabreauto-spdif
   - fsl,imx25-pdk-sgtl5000
   - fsl,imx53-cpuvo-sgtl5000
   - fsl,imx51-babbage-sgtl5000
@@ -54,6 +55,7 @@ properties:
   - fsl,imx6q-ventana-sgtl5000
   - fsl,imx6sl-evk-wm8962
   - fsl,imx6sx-sdb-mqs
+  - fsl,imx6sx-sdb-spdif
   - fsl,imx6sx-sdb-wm8962
   - fsl,imx7d-evk-wm8960
   - karo,tx53-audio-sgtl5000
@@ -65,6 +67,7 @@ properties:
   - fsl,imx-audio-sgtl5000
   - fsl,imx-audio-wm8960
   - fsl,imx-audio-wm8962
+  - fsl,imx-audio-spdif
   - items:
   - enum:
   - fsl,imx-audio-ac97
@@ -81,6 +84,7 @@ properties:
   - fsl,imx-audio-wm8960
   - fsl,imx-audio-wm8962
   - fsl,imx-audio-wm8958
+  - fsl,imx-audio-spdif
 
   model:
 $ref: /schemas/types.yaml#/definitions/string
@@ -93,8 +97,15 @@ properties:
   need to add ASRC support via DPCM.
 
   audio-codec:
-$ref: /schemas/types.yaml#/definitions/phandle
-description: The phandle of an audio codec
+$ref: /schemas/types.yaml#/definitions/phandle-array
+description: |
+  The phandle of an audio codec.
+  With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter,
+  spdif_receiver or both.
+minItems: 1
+maxItems: 2
+items:
+  maxItems: 1
 
   audio-cpu:
 $ref: /schemas/types.yaml#/definitions/phandle
@@ -150,8 +161,10 @@ properties:
 description: dai-link uses bit clock inversion.
 
   mclk-id:
-$ref: /schemas/types.yaml#/definitions/uint32
-description: main clock id, specific for each card configuration.
+$ref: /schemas/types.yaml#/definitions/uint32-array
+description: Main clock id for each codec, specific for each card 
configuration.
+minItems: 1
+maxItems: 2
 
   mux-int-port:
 $ref: /schemas/types.yaml#/definitions/uint32
@@ -195,3 +208,12 @@ examples:
  "AIN2L", "Line In Jack",
  "AIN2R", "Line In Jack";
 };
+
+  - |
+sound-spdif-asrc {
+  compatible = "fsl,imx-audio-spdif";
+  model = "spdif-asrc-audio";
+  audio-cpu = <&spdif>;
+  audio-asrc = <&easrc>;
+  audio-codec = <&spdifdit>, <&spdifdir>;
+};
-- 
2.34.1

[PATCHv5 7/9] ASoC: dt-bindings: imx-audio-spdif: remove binding

imx-audio-spdif was merged into the fsl-asoc-card driver, and therefore
removed.

Signed-off-by: Elinor Montmasson 
---
 .../bindings/sound/fsl,imx-audio-spdif.yaml   | 66 ---
 1 file changed, 66 deletions(-)
 delete mode 100644 
Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml

diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml 
b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
deleted file mode 100644
index 5fc543d02ecb..
--- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
+++ /dev/null
@@ -1,66 +0,0 @@
-# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
-%YAML 1.2

-$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml#
-$schema: http://devicetree.org/meta-schemas/core.yaml#
-
-title: Freescale i.MX audio complex with S/PDIF transceiver
-
-maintainers:
-  - Shengjiu Wang 
-
-properties:
-  compatible:
-oneOf:
-  - items:
-  - enum:
-  - fsl,imx-sabreauto-spdif
-  - fsl,imx6sx-sdb-spdif
-  - const: fsl,imx-audio-spdif
-  - enum:
-  - fsl,imx-audio-spdif
-
-  model:
-$ref: /schemas/types.yaml#/definitions/string
-description: User specified audio sound card name
-
-  spdif-controller:
-$ref: /schemas/types.yaml#/definitions/phandle
-description: The phandle of the i.MX S/PDIF controller
-
-  spdif-out:
-type: boolean
-description:
-  If present, the transmitting function of S/PDIF will be enabled,
-  indicating there's a physical S/PDIF out connector or jack on the
-  board or it's connecting to some other IP block, such as an HDMI
-  encoder or display-controller.
-
-  spdif-in:
-type: boolean
-description:
-  If present, the receiving function of S/PDIF will be enabled,
-  indicating there is a physical S/PDIF in connector/jack on the board.
-
-required:
-  - compatible
-  - model
-  - spdif-controller
-
-anyOf:
-  - required:
-  - spdif-in
-  - required:
-  - spdif-out
-
-additionalProperties: false
-
-examples:
-  - |
-sound-spdif {
-compatible = "fsl,imx-audio-spdif";
-model = "imx-spdif";
-spdif-controller = <&spdif>;
-spdif-out;
-spdif-in;
-};
-- 
2.34.1

[PATCHv5 8/9] arm64: dts: imx8m: update spdif sound card node properties

Following merge of imx-spdif driver into fsl-asoc-card:
* update properties to match those used by fsl-asoc-card.
* S/PDIF in/out dummy codecs must now be declared explicitly, add and
  use them.

These modifications were tested only on an imx8mn-evk board.

Signed-off-by: Elinor Montmasson 
---
 arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +---
 arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +---
 arch/arm64/boot/dts/freescale/imx8mq-evk.dts  | 24 +++
 3 files changed, 43 insertions(+), 11 deletions(-)

diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi 
b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi
index 90d1901df2b1..348855a41852 100644
--- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi
+++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi
@@ -180,12 +180,21 @@ cpu {
};
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "imx-spdif";
-   spdif-controller = <&spdif1>;
-   spdif-out;
-   spdif-in;
+   audio-cpu = <&spdif1>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
};
 };
 
diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi 
b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi
index 9e0259ddf4bc..6a47e09703a7 100644
--- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi
+++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi
@@ -124,12 +124,21 @@ sound-wm8524 {
"Line Out Jack", "LINEVOUTR";
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "imx-spdif";
-   spdif-controller = <&spdif1>;
-   spdif-out;
-   spdif-in;
+   audio-cpu = <&spdif1>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
};
 
sound-micfil {
diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts 
b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts
index 7507548cdb16..b953865f0b46 100644
--- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts
+++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts
@@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec {
};
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "imx-spdif";
-   spdif-controller = <&spdif1>;
-   spdif-out;
-   spdif-in;
+   audio-cpu = <&spdif1>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
+   };
+
+   hdmi_arc_in: hdmi-arc-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
};
 
sound-hdmi-arc {
compatible = "fsl,imx-audio-spdif";
model = "imx-hdmi-arc";
-   spdif-controller = <&spdif2>;
-   spdif-in;
+   audio-cpu = <&spdif2>;
+   audio-codec = <&hdmi_arc_in>;
};
 };
 
-- 
2.34.1

[PATCHv5 9/9] ARM: dts: imx6: update spdif sound card node properties

Following merge of imx-spdif driver into fsl-asoc-card:
* update properties to match those used by fsl-asoc-card.
* S/PDIF in/out dummy codecs must now be declared explicitly, add and
  use them.

These modifications were tested only on an imx8mn-evk board.

Signed-off-by: Elinor Montmasson 
---
 arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts| 15 ---
 arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts| 15 ---
 arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts   |  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ---
 arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi|  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi| 15 ---
 arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi|  9 +++--
 .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi|  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi  |  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi  |  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts|  9 +++--
 arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi |  9 +++--
 12 files changed, 104 insertions(+), 28 deletions(-)

diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts 
b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts
index 95b49fc83f7b..5c664c0f2169 100644
--- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts
+++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts
@@ -127,12 +127,21 @@ simple-audio-card,codec {
};
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "imx-spdif";
-   spdif-controller = <&spdif>;
-   spdif-out;
-   spdif-in;
+   audio-cpu = <&spdif>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
};
 };
 
diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts 
b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts
index a7d5693c5ab7..8491d656ef17 100644
--- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts
+++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts
@@ -111,12 +111,21 @@ simple-audio-card,codec {
};
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "imx-spdif";
-   spdif-controller = <&spdif>;
-   spdif-in;
-   spdif-out;
+   audio-cpu = <&spdif>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
};
 };
 
diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts 
b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts
index 7c298d9aa21e..ea9a98887c7b 100644
--- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts
+++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts
@@ -90,11 +90,16 @@ sound-sgtl5000 {
ssi-controller = <&ssi1>;
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
sound-spdif {
compatible = "fsl,imx-audio-spdif";
model = "On-board SPDIF";
-   spdif-controller = <&spdif>;
-   spdif-out;
+   audio-cpu = <&spdif>;
+   audio-codec = <&spdif_out>;
};
 };
 
diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi 
b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi
index ea40623d12e5..6f4546c59d38 100644
--- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi
+++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi
@@ -197,11 +197,20 @@ sound {
ssi-controller = <&ssi1>;
};
 
+   spdif_out: spdif-out {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dit";
+   };
+
+   spdif_in: spdif-in {
+   #sound-dai-cells = <0>;
+   compatible = "linux,spdif-dir";
+   };
+
sound_spdif: sound-spdif {
compatible = "fsl,imx-audio-spdif";
-   spdif-controller = <&spdif>;
-   spdif-in;
-   spdif-out;
+   audio-cpu = <&spdif>;
+   audio-codec = <&spdif_out>, <&spdif_in>;
model = "imx-spdif";
status = "disabled";
};
diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi 
b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi
index 3a46ade3b6bd..6aa6b152c3ae 100644
--- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi
+++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi
@@ -121,11 +121,16 @@ sound {
mux-

Re: [PATCH v2 0/2] powerpc: kexec fixes

On Fri, 10 May 2024 15:52:33 +0530, Sourabh Jain wrote:
> Patch series fixes two kexec issues.
> 
> 01/02: Update extra size calculation for kexec FDT to avoid kexec load
> failure due to FDT_ERR_NOSPACE while including CPU nodes added post
> boot and reserved memory ranges.
> 
> 02/02: Fix update_cpus_node/core_64.c function to include missing device
> nodes under /cpus node with device_type != "cpu".
> 
> [...]

Applied to powerpc/next.

[1/2] powerpc/kexec_file: fix extra size calculation for kexec FDT
  https://git.kernel.org/powerpc/c/0d3ff067331ef84e7e7f49537d768881042ed5ba
[2/2] powerpc/kexec_file: fix cpus node update to FDT
  https://git.kernel.org/powerpc/c/932bed41217059638c78a75411b7893b121d2162

cheers

Re: [PATCH v2 0/2] Fix doorbell emulation for v2 API on PPC

On Wed, 05 Jun 2024 17:09:08 +0530, Gautam Menghani wrote:
> Doorbell emulation for KVM on PAPR guests is broken as support for DPDES
> was not added in initial patch series [1].
> Add DPDES support and doorbell handling support for V2 API.
> 
> [1] lore.kernel.org/linuxppc-dev/20230914030600.16993-1-jniet...@gmail.com
> 
> Changes in v2:
> 1. Split DPDES support into its own patch
> 
> [...]

Applied to powerpc/topic/ppc-kvm.

[1/2] arch/powerpc/kvm: Add DPDES support in helper library for Guest state 
buffer
  https://git.kernel.org/powerpc/c/55dfb8bed6fe8bda390cc71cca878d11a9407099
[2/2] arch/powerpc/kvm: Fix doorbell emulation for v2 API
  https://git.kernel.org/powerpc/c/54ec2bd9e0173b75daf84675d07c56584f96564b

cheers

Re: [PATCH v9] arch/powerpc/kvm: Add support for reading VPA counters for pseries guests

On Mon, 20 May 2024 23:27:40 +0530, Gautam Menghani wrote:
> PAPR hypervisor has introduced three new counters in the VPA area of
> LPAR CPUs for KVM L2 guest (see [1] for terminology) observability - 2
> for context switches from host to guest and vice versa, and 1 counter
> for getting the total time spent inside the KVM guest. Add a tracepoint
> that enables reading the counters for use by ftrace/perf. Note that this
> tracepoint is only available for nestedv2 API (i.e, KVM on PowerVM).
> 
> [...]

Applied to powerpc/topic/ppc-kvm.

[1/1] arch/powerpc/kvm: Add support for reading VPA counters for pseries guests
  https://git.kernel.org/powerpc/c/e1f288d2f9c69bb8965db9fb99a19b58231a00dd

cheers

Re: [PATCH v2 0/8] KVM: PPC: Book3S HV: Nested guest migration fixes