[PATCH AUTOSEL 6.2 02/45] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds

[PATCH AUTOSEL 6.2 02/45] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds

[Sasha Levin]

From: Ravulapati Vishnu Vardhan Rao <quic_visr@quicinc.com>

[ Upstream commit e5e7e398f6bb7918dab0612eb6991f7bae95520d ]

When we run syzkaller we get below Out of Bound.
    "KASAN: slab-out-of-bounds Read in regcache_flat_read"

    Below is the backtrace of the issue:

    dump_backtrace+0x0/0x4c8
    show_stack+0x34/0x44
    dump_stack_lvl+0xd8/0x118
    print_address_description+0x30/0x2d8
    kasan_report+0x158/0x198
    __asan_report_load4_noabort+0x44/0x50
    regcache_flat_read+0x10c/0x110
    regcache_read+0xf4/0x180
    _regmap_read+0xc4/0x278
    _regmap_update_bits+0x130/0x290
    regmap_update_bits_base+0xc0/0x15c
    snd_soc_component_update_bits+0xa8/0x22c
    snd_soc_component_write_field+0x68/0xd4
    tx_macro_digital_mute+0xec/0x140

    Actually There is no need to have decimator with 32 bits.
    By limiting the variable with short type u8 issue is resolved.

Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr@quicinc.com>
Link: https://lore.kernel.org/r/20230304080702.609-1-quic_visr@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/lpass-tx-macro.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index 2449a2df66df0..8facdb922f076 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -242,7 +242,7 @@ enum {
 
 struct tx_mute_work {
 	struct tx_macro *tx;
-	u32 decimator;
+	u8 decimator;
 	struct delayed_work dwork;
 };
 
@@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx,
 	return 0;
 }
 
-static bool is_amic_enabled(struct snd_soc_component *component, int decimator)
+static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator)
 {
 	u16 adc_mux_reg, adc_reg, adc_n;
 
@@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w,
 			       struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	unsigned int decimator;
+	u8 decimator;
 	u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg;
 	u8 hpf_cut_off_freq;
 	int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS;
@@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_component *component = dai->component;
-	u32 decimator, sample_rate;
+	u32 sample_rate;
+	u8 decimator;
 	int tx_fs_rate;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
 
@@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
-	u16 decimator;
+	u8 decimator;
 
 	/* active decimator not set yet */
 	if (tx->active_decimator[dai->id] == -1)
-- 
2.39.2

[PATCH AUTOSEL 6.2 03/45] ASoC: Intel: avs: max98357a: Explicitly define codec format

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit d16c893425d07ada1fdd817ec06d322efcf69480 ]

max98357a is speaker codec configured in 48000/2/S16_LE format
regardless of front end format, so force it to be so.

Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230303134854.2277146-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/boards/max98357a.c | 22 ++++++++++++++++++++++
 1 file changed, 22 insertions(+)

diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index 921f42caf7e09..183123d08c5a3 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -8,6 +8,7 @@
 
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-acpi.h>
 #include <sound/soc-dapm.h>
@@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
 	{ "Spk", NULL, "Speaker" },
 };
 
+static int
+avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_max98357a_be_fixup;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
 	dl->dpcm_playback = 1;
-- 
2.39.2

[PATCH AUTOSEL 6.2 04/45] ASoC: Intel: avs: da7219: Explicitly define codec format

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit 61f368624fe4d0c25c6e9c917574b8ace51d776e ]

da7219 is headset codec configured in 48000/2/S24_LE format regardless
of front end format, so force it to be so.

Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230303134854.2277146-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/boards/da7219.c | 21 +++++++++++++++++++++
 1 file changed, 21 insertions(+)

diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index acd43b6108e99..1a1d572cc1d02 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL);
 }
 
+static int
+avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_da7219_be_fixup;
 	dl->init = avs_da7219_codec_init;
 	dl->exit = avs_da7219_codec_exit;
 	dl->nonatomic = 1;
-- 
2.39.2

[PATCH AUTOSEL 6.2 05/45] ASoC: Intel: avs: rt5682: Explicitly define codec format

[Sasha Levin]

From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>

[ Upstream commit d24dbc865c2bd5946bef62bb862a65df092dfc79 ]

rt5682 is headset codec configured in 48000/2/S24_LE format regardless
of front end format, so force it to be so.

Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230303134854.2277146-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/boards/rt5682.c | 22 ++++++++++++++++++++++
 1 file changed, 22 insertions(+)

diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 473e9fe5d0bf7..b2c2ba93dcb56 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = {
 	.hw_params = avs_rt5682_hw_params,
 };
 
+static int
+avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSPN to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->id = 0;
 	dl->init = avs_rt5682_codec_init;
 	dl->exit = avs_rt5682_codec_exit;
+	dl->be_hw_params_fixup = avs_rt5682_be_fixup;
 	dl->ops = &avs_rt5682_ops;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
-- 
2.39.2

[PATCH AUTOSEL 6.2 06/45] ASoC: Intel: avs: ssm4567: Remove nau8825 bits

[Sasha Levin]

From: Cezary Rojewski <cezary.rojewski@intel.com>

[ Upstream commit 933de2d127281731166cf2880fa1e23c5a0f7faa ]

Some of the nau8825 clock control got into the ssm4567, remove it.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230303134854.2277146-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/boards/ssm4567.c | 31 ----------------------------
 1 file changed, 31 deletions(-)

diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index c5db696127624..2b7f5ad92aca7 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -15,7 +15,6 @@
 #include <sound/soc-acpi.h>
 #include "../../../codecs/nau8825.h"
 
-#define SKL_NUVOTON_CODEC_DAI	"nau8825-hifi"
 #define SKL_SSM_CODEC_DAI	"ssm4567-hifi"
 
 static struct snd_soc_codec_conf card_codec_conf[] = {
@@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Right Speaker"),
 };
 
-static int
-platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event)
-{
-	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct snd_soc_card *card = dapm->card;
-	struct snd_soc_dai *codec_dai;
-	int ret;
-
-	codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI);
-	if (!codec_dai) {
-		dev_err(card->dev, "Codec dai not found\n");
-		return -EINVAL;
-	}
-
-	if (SND_SOC_DAPM_EVENT_ON(event)) {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
-					     SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	} else {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	}
-
-	return ret;
-}
-
 static const struct snd_soc_dapm_widget card_widgets[] = {
 	SND_SOC_DAPM_SPK("Left Speaker", NULL),
 	SND_SOC_DAPM_SPK("Right Speaker", NULL),
 	SND_SOC_DAPM_SPK("DP1", NULL),
 	SND_SOC_DAPM_SPK("DP2", NULL),
-	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control,
-			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route card_base_routes[] = {
-- 
2.39.2

[PATCH AUTOSEL 6.2 07/45] ASoC: Intel: avs: nau8825: Adjust clock control

[Sasha Levin]

From: Cezary Rojewski <cezary.rojewski@intel.com>

[ Upstream commit 6206b2e787da2ed567922c37bb588a44f6fb6705 ]

Internal clock shall be adjusted also in cases when DAPM event other
than 'ON' is triggered.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230303134854.2277146-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/avs/boards/nau8825.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index 6731d8a490767..49438a67a77c6 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co
 		return -EINVAL;
 	}
 
-	if (!SND_SOC_DAPM_EVENT_ON(event)) {
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
+					     SND_SOC_CLOCK_IN);
+	else
 		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0) {
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-			return ret;
-		}
-	}
+	if (ret < 0)
+		dev_err(card->dev, "Set sysclk failed: %d\n", ret);
 
-	return 0;
+	return ret;
 }
 
 static const struct snd_kcontrol_new card_controls[] = {
-- 
2.39.2

[PATCH AUTOSEL 6.2 14/45] ASoC: SOF: ipc3: Check for upper size limit for the received message

[Sasha Levin]

From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>

[ Upstream commit 989a3e4479177d0f4afab8be1960731bc0ffbbd0 ]

The sof_ipc3_rx_msg() checks for minimum size of a new rx message but it is
missing the check for upper limit.
Corrupted or compromised firmware might be able to take advantage of this
to cause out of bounds reads outside of the message area.

Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Curtis Malainey <curtis@malainey.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307114917.5124-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/ipc3.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
index 1fef4dcc09368..fde8af5a1f485 100644
--- a/sound/soc/sof/ipc3.c
+++ b/sound/soc/sof/ipc3.c
@@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev)
 		return;
 	}
 
-	if (hdr.size < sizeof(hdr)) {
-		dev_err(sdev->dev, "The received message size is invalid\n");
+	if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) {
+		dev_err(sdev->dev, "The received message size is invalid: %u\n",
+			hdr.size);
 		return;
 	}
 
-- 
2.39.2

[PATCH AUTOSEL 6.2 15/45] ASoC: SOF: ipc4-topology: Fix incorrect sample rate print unit

[Sasha Levin]

From: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>

[ Upstream commit 9e269e3aa9006440de639597079ee7140ef5b5f3 ]

This patch fixes the sample rate print unit from KHz to Hz.
E.g. 48000KHz becomes 48000Hz.

Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307110751.2053-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/ipc4-topology.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 59f4d42f9011e..65da1cf790d9c 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -155,7 +155,7 @@ static void sof_ipc4_dbg_audio_format(struct device *dev,
 	for (i = 0; i < num_format; i++, ptr = (u8 *)ptr + object_size) {
 		fmt = ptr;
 		dev_dbg(dev,
-			" #%d: %uKHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n",
+			" #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n",
 			i, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map,
 			fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg);
 	}
-- 
2.39.2

[PATCH AUTOSEL 6.2 16/45] ASoC: SOF: Intel: pci-tng: revert invalid bar size setting

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

[ Upstream commit ca09e2a351fbc7836ba9418304ff0c3e72addfe0 ]

The logic for the ioremap is to find the resource index 3 (IRAM) and
infer the BAR address by subtracting the IRAM offset. The BAR size
defined in hardware specifications is 2MB.

The commit 5947b2726beb6 ("ASoC: SOF: Intel: Check the bar size before
remapping") tried to find the BAR size by querying the resource length
instead of a pre-canned value, but by requesting the size for index 3
it only gets the size of the IRAM. That's obviously wrong and prevents
the probe from proceeding.

This commit attempted to fix an issue in a fuzzing/simulated
environment but created another on actual devices, so the best course
of action is to revert that change.

Reported-by: Ferry Toth <fntoth@gmail.com>
Tested-by: Ferry Toth <fntoth@gmail.com> (Intel Edison-Arduino)
Link: https://github.com/thesofproject/linux/issues/3901
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307095341.3222-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/pci-tng.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
index 5b2b409752c58..8c22a00266c06 100644
--- a/sound/soc/sof/intel/pci-tng.c
+++ b/sound/soc/sof/intel/pci-tng.c
@@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev)
 
 	/* LPE base */
 	base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET;
-	size = pci_resource_len(pci, desc->resindex_lpe_base);
-	if (size < PCI_BAR_SIZE) {
-		dev_err(sdev->dev, "error: I/O region is too small.\n");
-		return -ENODEV;
-	}
+	size = PCI_BAR_SIZE;
 
 	dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size);
 	sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size);
-- 
2.39.2

[PATCH AUTOSEL 6.2 17/45] ASoC: SOF: Intel: hda-dsp: harden D0i3 programming sequence

[Sasha Levin]

From: Rander Wang <rander.wang@intel.com>

[ Upstream commit 52a55779ed14792a150421339664193d6eb8e036 ]

Add delay between set and wait command according to hardware programming
sequence. Also add debug log to detect error.

Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307095453.3719-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/hda-dsp.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index b4eacae8564c8..6b2094f74c9c0 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -399,6 +399,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset,
 			    SOF_HDA_VS_D0I3C_I3, value);
 
+	/*
+	 * The value written to the D0I3C::I3 bit may not be taken into account immediately.
+	 * A delay is recommended before checking if D0I3C::CIP is cleared
+	 */
+	usleep_range(30, 40);
+
 	/* Wait for cmd in progress to be cleared before exiting the function */
 	ret = hda_dsp_wait_d0i3c_done(sdev);
 	if (ret < 0) {
@@ -407,6 +413,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	}
 
 	reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset);
+	/* Confirm d0i3 state changed with paranoia check */
+	if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) {
+		dev_err(sdev->dev, "failed to update D0I3C!\n");
+		return -EIO;
+	}
+
 	trace_sof_intel_D0I3C_updated(sdev, reg);
 
 	return 0;
-- 
2.39.2

[PATCH AUTOSEL 6.2 18/45] ASoC: SOF: Intel: hda-ctrl: re-add sleep after entering and exiting reset

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

[ Upstream commit 8bac40b8ed17ab1be9133e9620f65fae80262b7e ]

This reverts commit a09d82ce0a867 ("ASoC: SOF: Intel: hda-ctrl: remove
useless sleep")

It was a mistake to remove those delays, in light of comments in the
HDaudio spec captured in snd_hdac_bus_reset_link() that the codec
needs time for its initialization and PLL lock.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307095412.3416-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/intel/hda-ctrl.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index 3aea36c077c9d..f3bdeba284122 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -196,12 +196,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
 		goto err;
 	}
 
+	usleep_range(500, 1000);
+
 	/* exit HDA controller reset */
 	ret = hda_dsp_ctrl_link_reset(sdev, false);
 	if (ret < 0) {
 		dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
 		goto err;
 	}
+	usleep_range(1000, 1200);
 
 	hda_codec_detect_mask(sdev);
 
-- 
2.39.2

[PATCH AUTOSEL 6.2 19/45] ASoC: SOF: IPC4: update gain ipc msg definition to align with fw

[Sasha Levin]

From: Rander Wang <rander.wang@intel.com>

[ Upstream commit e45cd86c3a78bfb9875a5eb8ab5dab459b59bbe2 ]

Recent firmware changes modified the curve duration from 32 to 64 bits,
which breaks volume ramps. A simple solution would be to change the
definition, but unfortunately the ASoC topology framework only supports
up to 32 bit tokens.

This patch suggests breaking the 64 bit value in low and high parts, with
only the low-part extracted from topology and high-part only zeroes. Since
the curve duration is represented in hundred of nanoseconds, we can still
represent a 400s ramp, which is just fine. The defacto ABI change has no
effect on existing users since the IPC4 firmware has not been released just
yet.

Link: https://github.com/thesofproject/linux/issues/4026

Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230307110656.1816-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/ipc4-control.c  | 3 ++-
 sound/soc/sof/ipc4-topology.c | 4 ++--
 sound/soc/sof/ipc4-topology.h | 6 ++++--
 3 files changed, 8 insertions(+), 5 deletions(-)

diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
index 0d5a578c34962..7442ec1c5a4d4 100644
--- a/sound/soc/sof/ipc4-control.c
+++ b/sound/soc/sof/ipc4-control.c
@@ -84,7 +84,8 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge
 		}
 
 		/* set curve type and duration from topology */
-		data.curve_duration = gain->data.curve_duration;
+		data.curve_duration_l = gain->data.curve_duration_l;
+		data.curve_duration_h = gain->data.curve_duration_h;
 		data.curve_type = gain->data.curve_type;
 
 		msg->data_ptr = &data;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 65da1cf790d9c..6da6137fa2cbc 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -107,7 +107,7 @@ static const struct sof_topology_token gain_tokens[] = {
 		get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)},
 	{SOF_TKN_GAIN_RAMP_DURATION,
 		SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
-		offsetof(struct sof_ipc4_gain_data, curve_duration)},
+		offsetof(struct sof_ipc4_gain_data, curve_duration_l)},
 	{SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD,
 		get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)},
 };
@@ -670,7 +670,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget)
 
 	dev_dbg(scomp->dev,
 		"pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x, cpc %d\n",
-		swidget->widget->name, gain->data.curve_type, gain->data.curve_duration,
+		swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l,
 		gain->data.init_val, gain->base_config.cpc);
 
 	ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg);
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index 0aa87a8add5d3..edf1638221a4b 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -217,14 +217,16 @@ struct sof_ipc4_control_data {
  * @init_val: Initial value
  * @curve_type: Curve type
  * @reserved: reserved for future use
- * @curve_duration: Curve duration
+ * @curve_duration_l: Curve duration low part
+ * @curve_duration_h: Curve duration high part
  */
 struct sof_ipc4_gain_data {
 	uint32_t channels;
 	uint32_t init_val;
 	uint32_t curve_type;
 	uint32_t reserved;
-	uint32_t curve_duration;
+	uint32_t curve_duration_l;
+	uint32_t curve_duration_h;
 } __aligned(8);
 
 /**
-- 
2.39.2

[PATCH AUTOSEL 6.2 20/45] ASoC: hdmi-codec: only startup/shutdown on supported streams

[Sasha Levin]

From: Emil Abildgaard Svendsen <EMAS@bang-olufsen.dk>

[ Upstream commit e041a2a550582106cba6a7c862c90dfc2ad14492 ]

Currently only one stream is supported. This isn't usally a problem
until you have a multi codec audio card. Because the audio card will run
startup and shutdown on both capture and playback streams. So if your
hdmi-codec only support either playback or capture. Then ALSA can't open
for playback and capture.

This patch will ignore if startup and shutdown are called with a non
supported stream. Thus, allowing an audio card like this:

           +-+
 cpu1 <--@-| |-> codec1 (HDMI-CODEC)
           | |<- codec2 (NOT HDMI-CODEC)
           +-+

Signed-off-by: Emil Svendsen <emas@bang-olufsen.dk>
Link: https://lore.kernel.org/r/20230309065432.4150700-2-emas@bang-olufsen.dk
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/hdmi-codec.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 74cbbe16f9aec..a22f2ec95901f 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
 	int ret = 0;
 
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return 0;
+
 	mutex_lock(&hcp->lock);
 	if (hcp->busy) {
 		dev_err(dai->dev, "Only one simultaneous stream supported!\n");
@@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
+
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return;
 
 	hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
 	hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data);
-- 
2.39.2

[PATCH AUTOSEL 6.2 30/45] ALSA: asihpi: check pao in control_message()

[Sasha Levin]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

[ Upstream commit 9026c0bf233db53b86f74f4c620715e94eb32a09 ]

control_message() might be called with pao = NULL.
Here indicates control_message() as sample.

(B)	static void control_message(struct hpi_adapter_obj *pao, ...)
	{                                                   ^^^
		struct hpi_hw_obj *phw = pao->priv;
		...                      ^^^
	}

(A)	void _HPI_6205(struct hpi_adapter_obj *pao, ...)
	{                                      ^^^
		...
		case HPI_OBJ_CONTROL:
(B)			control_message(pao, phm, phr);
			break;          ^^^
		...
	}

	void HPI_6205(...)
	{
		...
(A)		_HPI_6205(NULL, phm, phr);
		...       ^^^^
	}

Therefore, We will get too many warning via cppcheck, like below

	sound/pci/asihpi/hpi6205.c:238:27: warning: Possible null pointer dereference: pao [nullPointer]
		 struct hpi_hw_obj *phw = pao->priv;
		                          ^
	sound/pci/asihpi/hpi6205.c:433:13: note: Calling function '_HPI_6205', 1st argument 'NULL' value is 0
		  _HPI_6205(NULL, phm, phr);
		            ^
	sound/pci/asihpi/hpi6205.c:401:20: note: Calling function 'control_message', 1st argument 'pao' value is 0
	   control_message(pao, phm, phr);
	                   ^
Set phr->error like many functions doing, and don't call _HPI_6205()
with NULL.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ttypeaqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/pci/asihpi/hpi6205.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 27e11b5f70b97..c7d7eff86727f 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -430,7 +430,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr)
 		pao = hpi_find_adapter(phm->adapter_index);
 	} else {
 		/* subsys messages don't address an adapter */
-		_HPI_6205(NULL, phm, phr);
+		phr->error = HPI_ERROR_INVALID_OBJ_INDEX;
 		return;
 	}
 
-- 
2.39.2

[PATCH AUTOSEL 6.2 31/45] ALSA: hda/ca0132: fixup buffer overrun at tuning_ctl_set()

[Sasha Levin]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

[ Upstream commit 98e5eb110095ec77cb6d775051d181edbf9cd3cf ]

tuning_ctl_set() might have buffer overrun at (X) if it didn't break
from loop by matching (A).

	static int tuning_ctl_set(...)
	{
		for (i = 0; i < TUNING_CTLS_COUNT; i++)
(A)			if (nid == ca0132_tuning_ctls[i].nid)
				break;

		snd_hda_power_up(...);
(X)		dspio_set_param(..., ca0132_tuning_ctls[i].mid, ...);
		snd_hda_power_down(...);                ^

		return 1;
	}

We will get below error by cppcheck

	sound/pci/hda/patch_ca0132.c:4229:2: note: After for loop, i has value 12
	 for (i = 0; i < TUNING_CTLS_COUNT; i++)
	 ^
	sound/pci/hda/patch_ca0132.c:4234:43: note: Array index out of bounds
	 dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20,
	                                           ^
This patch cares non match case.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sfe9eap7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/pci/hda/patch_ca0132.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index acde4cd58785e..099722ebaed83 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -4228,8 +4228,10 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid,
 
 	for (i = 0; i < TUNING_CTLS_COUNT; i++)
 		if (nid == ca0132_tuning_ctls[i].nid)
-			break;
+			goto found;
 
+	return -EINVAL;
+found:
 	snd_hda_power_up(codec);
 	dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20,
 			ca0132_tuning_ctls[i].req,
-- 
2.39.2