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Code Issues Releases Wiki Activity

ASoC: Fixes for v6.4 

More fixes that came in since the merge window, the bulk of which are
 for the SOF code, I suspect as a result of the wide usage, active
 development and large code size rather than huge quality problems.
 
 There's also a couple of MAINTAINERS updates and some new device quirks.
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Merge tag 'asoc-fix-v6.4-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.4

More fixes that came in since the merge window, the bulk of which are
for the SOF code, I suspect as a result of the wide usage, active
development and large code size rather than huge quality problems.

There's also a couple of MAINTAINERS updates and some new device quirks.
This commit is contained in:
Takashi Iwai 2023-05-16 20:11:50 +02:00
parent 9dc68a4fe7 9be0b3a007
commit cc638dba23
26 changed files with 279 additions and 84 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
Documentation/devicetree/bindings/sound/tas2562.yaml
View File

@ -55,7 +55,9 @@ properties:
description: TDM TX current sense time slot.
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -72,7 +74,7 @@ examples:
codec: codec@4c {
compatible = "ti,tas2562";
reg = <0x4c>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
shutdown-gpios = <&gpio1 15 0>;

6
Documentation/devicetree/bindings/sound/tas2770.yaml
View File

@ -57,7 +57,9 @@ properties:
- 1 # Falling edge
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -74,7 +76,7 @@ examples:
codec: codec@41 {
compatible = "ti,tas2770";
reg = <0x41>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
reset-gpio = <&gpio1 15 0>;

6
Documentation/devicetree/bindings/sound/tas27xx.yaml
View File

@ -50,7 +50,9 @@ properties:
description: TDM TX voltage sense time slot.
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -67,7 +69,7 @@ examples:
codec: codec@38 {
compatible = "ti,tas2764";
reg = <0x38>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
reset-gpios = <&gpio1 15 0>;

4
MAINTAINERS
View File

@ -4921,7 +4921,6 @@ F: drivers/media/cec/i2c/ch7322.c
CIRRUS LOGIC AUDIO CODEC DRIVERS
M: James Schulman <james.schulman@cirrus.com>
M: David Rhodes <david.rhodes@cirrus.com>
M: Lucas Tanure <tanureal@opensource.cirrus.com>
M: Richard Fitzgerald <rf@opensource.cirrus.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: patches@opensource.cirrus.com
@ -18568,10 +18567,9 @@ F: Documentation/admin-guide/LSM/SafeSetID.rst
F: security/safesetid/
SAMSUNG AUDIO (ASoC) DRIVERS
M: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
M: Sylwester Nawrocki <s.nawrocki@samsung.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
S: Maintained
B: mailto:linux-samsung-soc@vger.kernel.org
F: Documentation/devicetree/bindings/sound/samsung*
F: sound/soc/samsung/

14
include/sound/hda-mlink.h
View File

@ -44,6 +44,9 @@ int hdac_bus_eml_sdw_power_down_unlocked(struct hdac_bus *bus, int sublink);
int hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num);
int hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
int channel_mask, int stream_id, int dir);
void hda_bus_ml_put_all(struct hdac_bus *bus);
void hda_bus_ml_reset_losidv(struct hdac_bus *bus);
int hda_bus_ml_resume(struct hdac_bus *bus);
@ -51,6 +54,7 @@ int hda_bus_ml_suspend(struct hdac_bus *bus);
struct hdac_ext_link *hdac_bus_eml_ssp_get_hlink(struct hdac_bus *bus);
struct hdac_ext_link *hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus);
struct hdac_ext_link *hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus);
struct mutex *hdac_bus_eml_get_mutex(struct hdac_bus *bus, bool alt, int elid);
@ -144,6 +148,13 @@ hdac_bus_eml_sdw_power_down_unlocked(struct hdac_bus *bus, int sublink) { return
static inline int
hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num) { return 0; }
static inline int
hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
int channel_mask, int stream_id, int dir)
{
return 0;
}
static inline void hda_bus_ml_put_all(struct hdac_bus *bus) { }
static inline void hda_bus_ml_reset_losidv(struct hdac_bus *bus) { }
static inline int hda_bus_ml_resume(struct hdac_bus *bus) { return 0; }
@ -155,6 +166,9 @@ hdac_bus_eml_ssp_get_hlink(struct hdac_bus *bus) { return NULL; }
static inline struct hdac_ext_link *
hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus) { return NULL; }
static inline struct hdac_ext_link *
hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus) { return NULL; }
static inline struct mutex *
hdac_bus_eml_get_mutex(struct hdac_bus *bus, bool alt, int elid) { return NULL; }

3
include/uapi/sound/sof/tokens.h
View File

@ -183,7 +183,7 @@
#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_INTERLEAVING_STYLE 1906
#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG 1907
#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_SAMPLE_TYPE 1908
#define SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX 1909
#define SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX 1909
/* intentional token numbering discontinuity, reserved for future use */
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_RATE 1930
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_BIT_DEPTH 1931
@ -194,6 +194,7 @@
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_INTERLEAVING_STYLE 1936
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG 1937
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_SAMPLE_TYPE 1938
#define SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX 1939
/* intentional token numbering discontinuity, reserved for future use */
#define SOF_TKN_CAVS_AUDIO_FORMAT_IBS 1970
#define SOF_TKN_CAVS_AUDIO_FORMAT_OBS 1971

7
sound/soc/amd/yc/acp6x-mach.c
View File

@ -311,6 +311,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "8A22"),
}
},
{
.driver_data = &acp6x_card,
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "System76"),
DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"),
}
},
{}
};

3
sound/soc/codecs/cs35l56.c
View File

@ -852,10 +852,11 @@ static void cs35l56_dsp_work(struct work_struct *work)
*/
if (cs35l56->sdw_peripheral) {
cs35l56->sdw_irq_no_unmask = true;
cancel_work_sync(&cs35l56->sdw_irq_work);
flush_work(&cs35l56->sdw_irq_work);
sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0);
sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1);
sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF);
flush_work(&cs35l56->sdw_irq_work);
}
ret = cs35l56_mbox_send(cs35l56, CS35L56_MBOX_CMD_SHUTDOWN);

15
sound/soc/codecs/ssm2602.c
View File

@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = {
{ .reg = 0x09, .def = 0x0000 }
};
/*
* ssm2602 register patch
* Workaround for playback distortions after power up: activates digital
* core, and then powers on output, DAC, and whole chip at the same time
*/
static const struct reg_sequence ssm2602_patch[] = {
{ SSM2602_ACTIVE, 0x01 },
{ SSM2602_PWR, 0x07 },
{ SSM2602_RESET, 0x00 },
};
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
@ -598,6 +610,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component)
return ret;
}
regmap_register_patch(ssm2602->regmap, ssm2602_patch,
ARRAY_SIZE(ssm2602_patch));
/* set the update bits */
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);

4
sound/soc/dwc/dwc-i2s.c
View File

@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
/* Error Handling: TX */
if (isr[i] & ISR_TXFO) {
dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
/* Error Handling: TX */
if (isr[i] & ISR_RXFO) {
dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
}

14
sound/soc/fsl/fsl_micfil.c
View File

@ -1159,7 +1159,7 @@ static int fsl_micfil_probe(struct platform_device *pdev)
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
if (ret) {
dev_err(&pdev->dev, "failed to pcm register\n");
return ret;
goto err_pm_disable;
}
fsl_micfil_dai.capture.formats = micfil->soc->formats;
@ -1169,9 +1169,20 @@ static int fsl_micfil_probe(struct platform_device *pdev)
if (ret) {
dev_err(&pdev->dev, "failed to register component %s\n",
fsl_micfil_component.name);
goto err_pm_disable;
}
return ret;
err_pm_disable:
pm_runtime_disable(&pdev->dev);
return ret;
}
static void fsl_micfil_remove(struct platform_device *pdev)
{
pm_runtime_disable(&pdev->dev);
}
static int __maybe_unused fsl_micfil_runtime_suspend(struct device *dev)
@ -1232,6 +1243,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = {
static struct platform_driver fsl_micfil_driver = {
.probe = fsl_micfil_probe,
.remove_new = fsl_micfil_remove,
.driver = {
.name = "fsl-micfil-dai",
.pm = &fsl_micfil_pm_ops,

54
sound/soc/jz4740/jz4740-i2s.c
View File

@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate,
unsigned long i2sdiv_max)
{
unsigned long div, rate1, rate2, err1, err2;
div = mclk / (64 * rate);
if (div == 0)
div = 1;
rate1 = mclk / (64 * div);
rate2 = mclk / (64 * (div + 1));
err1 = abs(rate1 - rate);
err2 = abs(rate2 - rate);
/*
* Choose the divider that produces the smallest error in the
* output rate and reject dividers with a 5% or higher error.
* In the event that both dividers are outside the acceptable
* error margin, reject the rate to prevent distorted audio.
* (The number 5% is arbitrary.)
*/
if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20)
return div;
if (div < i2sdiv_max && err2 < rate/20)
return div + 1;
return -EINVAL;
}
static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
struct regmap_field *div_field;
unsigned long i2sdiv_max;
unsigned int sample_size;
uint32_t ctrl;
int div;
uint32_t ctrl, conf;
int div = 1;
regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl);
div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
div_field = i2s->field_i2sdiv_playback;
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb,
i2s->soc_info->field_i2sdiv_playback.lsb);
} else {
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE;
ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size);
div_field = i2s->field_i2sdiv_capture;
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb,
i2s->soc_info->field_i2sdiv_capture.lsb);
}
/*
* Only calculate I2SDIV if we're supplying the bit or frame clock.
* If the codec is supplying both clocks then the divider output is
* unused, and we don't want it to limit the allowed sample rates.
*/
if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) {
div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s),
params_rate(params), i2sdiv_max);
if (div < 0)
return div;
}
regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl);

6
sound/soc/mediatek/mt8186/mt8186-afe-clk.c
View File

@ -644,9 +644,3 @@ int mt8186_init_clock(struct mtk_base_afe *afe)
return 0;
}
void mt8186_deinit_clock(void *priv)
{
struct mtk_base_afe *afe = priv;
mt8186_audsys_clk_unregister(afe);
}

1
sound/soc/mediatek/mt8186/mt8186-afe-clk.h
View File

@ -81,7 +81,6 @@ enum {
struct mtk_base_afe;
int mt8186_set_audio_int_bus_parent(struct mtk_base_afe *afe, int clk_id);
int mt8186_init_clock(struct mtk_base_afe *afe);
void mt8186_deinit_clock(void *priv);
int mt8186_afe_enable_cgs(struct mtk_base_afe *afe);
void mt8186_afe_disable_cgs(struct mtk_base_afe *afe);
int mt8186_afe_enable_clock(struct mtk_base_afe *afe);

4
sound/soc/mediatek/mt8186/mt8186-afe-pcm.c
View File

@ -2848,10 +2848,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev)
return ret;
}
ret = devm_add_action_or_reset(dev, mt8186_deinit_clock, (void *)afe);
if (ret)
return ret;
/* init memif */
afe->memif_32bit_supported = 0;
afe->memif_size = MT8186_MEMIF_NUM;

46
sound/soc/mediatek/mt8186/mt8186-audsys-clk.c
View File

@ -84,6 +84,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
GATE_AUD2(CLK_AUD_ETDM_OUT1_BCLK, "aud_etdm_out1_bclk", "top_audio", 24),
};
static void mt8186_audsys_clk_unregister(void *data)
{
struct mtk_base_afe *afe = data;
struct mt8186_afe_private *afe_priv = afe->platform_priv;
struct clk *clk;
struct clk_lookup *cl;
int i;
if (!afe_priv)
return;
for (i = 0; i < CLK_AUD_NR_CLK; i++) {
cl = afe_priv->lookup[i];
if (!cl)
continue;
clk = cl->clk;
clk_unregister_gate(clk);
clkdev_drop(cl);
}
}
int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
{
struct mt8186_afe_private *afe_priv = afe->platform_priv;
@ -124,27 +147,6 @@ int mt8186_audsys_clk_register(struct mtk_base_afe *afe)
afe_priv->lookup[i] = cl;
}
return 0;
return devm_add_action_or_reset(afe->dev, mt8186_audsys_clk_unregister, afe);
}
void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe)
{
struct mt8186_afe_private *afe_priv = afe->platform_priv;
struct clk *clk;
struct clk_lookup *cl;
int i;
if (!afe_priv)
return;
for (i = 0; i < CLK_AUD_NR_CLK; i++) {
cl = afe_priv->lookup[i];
if (!cl)
continue;
clk = cl->clk;
clk_unregister_gate(clk);
clkdev_drop(cl);
}
}

1
sound/soc/mediatek/mt8186/mt8186-audsys-clk.h
View File

@ -10,6 +10,5 @@
#define _MT8186_AUDSYS_CLK_H_
int mt8186_audsys_clk_register(struct mtk_base_afe *afe);
void mt8186_audsys_clk_unregister(struct mtk_base_afe *afe);
#endif

7
sound/soc/sof/amd/acp-ipc.c
View File

@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp
acp_mailbox_read(sdev, offset, p, sz);
} else {
struct snd_pcm_substream *substream = sps->substream;
struct acp_dsp_stream *stream = substream->runtime->private_data;
struct acp_dsp_stream *stream;
if (!substream || !substream->runtime)
return -ESTRPIPE;
stream = substream->runtime->private_data;
if (!stream)
return -ESTRPIPE;

4
sound/soc/sof/debug.c
View File

@ -438,8 +438,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg)
/* should we prevent DSP entering D3 ? */
if (!sdev->ipc_dump_printed)
dev_info(sdev->dev,
"preventing DSP entering D3 state to preserve context\n");
pm_runtime_get_noresume(sdev->dev);
"Attempting to prevent DSP from entering D3 state to preserve context\n");
pm_runtime_get_if_in_use(sdev->dev);
}
/* dump vital information to the logs */

96
sound/soc/sof/intel/hda-mlink.c
View File

@ -19,6 +19,9 @@
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_MLINK)
/* worst-case number of sublinks is used for sublink refcount array allocation only */
#define HDAML_MAX_SUBLINKS (AZX_ML_LCTL_CPA_SHIFT - AZX_ML_LCTL_SPA_SHIFT)
/**
* struct hdac_ext2_link - HDAudio extended+alternate link
*
@ -33,6 +36,7 @@
* @leptr: extended link pointer
* @eml_lock: mutual exclusion to access shared registers e.g. CPA/SPA bits
* in LCTL register
* @sublink_ref_count: array of refcounts, required to power-manage sublinks independently
* @base_ptr: pointer to shim/ip/shim_vs space
* @instance_offset: offset between each of @slcount instances managed by link
* @shim_offset: offset to SHIM register base
@ -53,6 +57,7 @@ struct hdac_ext2_link {
u32 leptr;
struct mutex eml_lock; /* prevent concurrent access to e.g. CPA/SPA */
int sublink_ref_count[HDAML_MAX_SUBLINKS];
/* internal values computed from LCAP contents */
void __iomem *base_ptr;
@ -68,6 +73,7 @@ struct hdac_ext2_link {
#define AZX_REG_SDW_SHIM_OFFSET 0x0
#define AZX_REG_SDW_IP_OFFSET 0x100
#define AZX_REG_SDW_VS_SHIM_OFFSET 0x6000
#define AZX_REG_SDW_SHIM_PCMSyCM(y) (0x16 + 0x4 * (y))
/* only one instance supported */
#define AZX_REG_INTEL_DMIC_SHIM_OFFSET 0x0
@ -91,7 +97,7 @@ struct hdac_ext2_link {
*/
static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
void __iomem *ml_addr, int link_idx)
void __iomem *remap_addr, void __iomem *ml_addr, int link_idx)
{
struct hdac_ext_link *hlink = &h2link->hext_link;
u32 base_offset;
@ -126,15 +132,16 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
link_idx, h2link->slcount);
/* find IP ID and offsets */
h2link->leptr = readl(hlink->ml_addr + AZX_REG_ML_LEPTR);
h2link->leptr = readl(ml_addr + AZX_REG_ML_LEPTR);
h2link->elid = FIELD_GET(AZX_REG_ML_LEPTR_ID, h2link->leptr);
base_offset = FIELD_GET(AZX_REG_ML_LEPTR_PTR, h2link->leptr);
h2link->base_ptr = hlink->ml_addr + base_offset;
h2link->base_ptr = remap_addr + base_offset;
switch (h2link->elid) {
case AZX_REG_ML_LEPTR_ID_SDW:
h2link->instance_offset = AZX_REG_SDW_INSTANCE_OFFSET;
h2link->shim_offset = AZX_REG_SDW_SHIM_OFFSET;
h2link->ip_offset = AZX_REG_SDW_IP_OFFSET;
h2link->shim_vs_offset = AZX_REG_SDW_VS_SHIM_OFFSET;
@ -149,6 +156,7 @@ static int hdaml_lnk_enum(struct device *dev, struct hdac_ext2_link *h2link,
link_idx, base_offset);
break;
case AZX_REG_ML_LEPTR_ID_INTEL_SSP:
h2link->instance_offset = AZX_REG_INTEL_SSP_INSTANCE_OFFSET;
h2link->shim_offset = AZX_REG_INTEL_SSP_SHIM_OFFSET;
h2link->ip_offset = AZX_REG_INTEL_SSP_IP_OFFSET;
h2link->shim_vs_offset = AZX_REG_INTEL_SSP_VS_SHIM_OFFSET;
@ -333,6 +341,21 @@ static void hdaml_link_set_lsdiid(u32 __iomem *lsdiid, int dev_num)
writel(val, lsdiid);
}
static void hdaml_shim_map_stream_ch(u16 __iomem *pcmsycm, int lchan, int hchan,
int stream_id, int dir)
{
u16 val;
val = readw(pcmsycm);
u16p_replace_bits(&val, lchan, GENMASK(3, 0));
u16p_replace_bits(&val, hchan, GENMASK(7, 4));
u16p_replace_bits(&val, stream_id, GENMASK(13, 8));
u16p_replace_bits(&val, dir, BIT(15));
writew(val, pcmsycm);
}
static void hdaml_lctl_offload_enable(u32 __iomem *lctl, bool enable)
{
u32 val = readl(lctl);
@ -364,7 +387,7 @@ static int hda_ml_alloc_h2link(struct hdac_bus *bus, int index)
hlink->bus = bus;
hlink->ml_addr = bus->mlcap + AZX_ML_BASE + (AZX_ML_INTERVAL * index);
ret = hdaml_lnk_enum(bus->dev, h2link, hlink->ml_addr, index);
ret = hdaml_lnk_enum(bus->dev, h2link, bus->remap_addr, hlink->ml_addr, index);
if (ret < 0) {
kfree(h2link);
return ret;
@ -641,8 +664,13 @@ static int hdac_bus_eml_power_up_base(struct hdac_bus *bus, bool alt, int elid,
if (eml_lock)
mutex_lock(&h2link->eml_lock);
if (++hlink->ref_count > 1)
goto skip_init;
if (!alt) {
if (++hlink->ref_count > 1)
goto skip_init;
} else {
if (++h2link->sublink_ref_count[sublink] > 1)
goto skip_init;
}
ret = hdaml_link_init(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
@ -684,9 +712,13 @@ static int hdac_bus_eml_power_down_base(struct hdac_bus *bus, bool alt, int elid
if (eml_lock)
mutex_lock(&h2link->eml_lock);
if (--hlink->ref_count > 0)
goto skip_shutdown;
if (!alt) {
if (--hlink->ref_count > 0)
goto skip_shutdown;
} else {
if (--h2link->sublink_ref_count[sublink] > 0)
goto skip_shutdown;
}
ret = hdaml_link_shutdown(hlink->ml_addr + AZX_REG_ML_LCTL, sublink);
skip_shutdown:
@ -740,6 +772,40 @@ int hdac_bus_eml_sdw_set_lsdiid(struct hdac_bus *bus, int sublink, int dev_num)
return 0;
} EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_set_lsdiid, SND_SOC_SOF_HDA_MLINK);
/*
* the 'y' parameter comes from the PCMSyCM hardware register naming. 'y' refers to the
* PDI index, i.e. the FIFO used for RX or TX
*/
int hdac_bus_eml_sdw_map_stream_ch(struct hdac_bus *bus, int sublink, int y,
int channel_mask, int stream_id, int dir)
{
struct hdac_ext2_link *h2link;
u16 __iomem *pcmsycm;
u16 val;
h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
if (!h2link)
return -ENODEV;
pcmsycm = h2link->base_ptr + h2link->shim_offset +
h2link->instance_offset * sublink +
AZX_REG_SDW_SHIM_PCMSyCM(y);
mutex_lock(&h2link->eml_lock);
hdaml_shim_map_stream_ch(pcmsycm, 0, hweight32(channel_mask),
stream_id, dir);
mutex_unlock(&h2link->eml_lock);
val = readw(pcmsycm);
dev_dbg(bus->dev, "channel_mask %#x stream_id %d dir %d pcmscm %#x\n",
channel_mask, stream_id, dir, val);
return 0;
} EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_map_stream_ch, SND_SOC_SOF_HDA_MLINK);
void hda_bus_ml_put_all(struct hdac_bus *bus)
{
struct hdac_ext_link *hlink;
@ -836,6 +902,18 @@ struct hdac_ext_link *hdac_bus_eml_dmic_get_hlink(struct hdac_bus *bus)
}
EXPORT_SYMBOL_NS(hdac_bus_eml_dmic_get_hlink, SND_SOC_SOF_HDA_MLINK);
struct hdac_ext_link *hdac_bus_eml_sdw_get_hlink(struct hdac_bus *bus)
{
struct hdac_ext2_link *h2link;
h2link = find_ext2_link(bus, true, AZX_REG_ML_LEPTR_ID_SDW);
if (!h2link)
return NULL;
return &h2link->hext_link;
}
EXPORT_SYMBOL_NS(hdac_bus_eml_sdw_get_hlink, SND_SOC_SOF_HDA_MLINK);
int hdac_bus_eml_enable_offload(struct hdac_bus *bus, bool alt, int elid, bool enable)
{
struct hdac_ext2_link *h2link;

7
sound/soc/sof/ipc3-topology.c
View File

@ -2103,10 +2103,13 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
* For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is.
*/
if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
/* Clear stale command */
config->flags &= ~SOF_DAI_CONFIG_FLAGS_CMD_MASK;
config->flags |= flags;
else
} else {
config->flags = flags;
}
/* only send the IPC if the widget is set up in the DSP */
if (swidget->use_count > 0) {

4
sound/soc/sof/ipc4-topology.c
View File

@ -59,7 +59,7 @@ static const struct sof_topology_token ipc4_in_audio_format_tokens[] = {
audio_fmt.interleaving_style)},
{SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
{SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
{SOF_TKN_CAVS_AUDIO_FORMAT_INPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, pin_index)},
{SOF_TKN_CAVS_AUDIO_FORMAT_IBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, buffer_size)},
@ -79,7 +79,7 @@ static const struct sof_topology_token ipc4_out_audio_format_tokens[] = {
audio_fmt.interleaving_style)},
{SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, audio_fmt.fmt_cfg)},
{SOF_TKN_CAVS_AUDIO_FORMAT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
{SOF_TKN_CAVS_AUDIO_FORMAT_OUTPUT_PIN_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, pin_index)},
{SOF_TKN_CAVS_AUDIO_FORMAT_OBS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
offsetof(struct sof_ipc4_pin_format, buffer_size)},

17
sound/soc/sof/pcm.c
View File

@ -643,16 +643,17 @@ static int sof_pcm_probe(struct snd_soc_component *component)
"%s/%s",
plat_data->tplg_filename_prefix,
plat_data->tplg_filename);
if (!tplg_filename)
return -ENOMEM;
ret = snd_sof_load_topology(component, tplg_filename);
if (ret < 0) {
dev_err(component->dev, "error: failed to load DSP topology %d\n",
ret);
return ret;
if (!tplg_filename) {
ret = -ENOMEM;
goto pm_error;
}
ret = snd_sof_load_topology(component, tplg_filename);
if (ret < 0)
dev_err(component->dev, "error: failed to load DSP topology %d\n",
ret);
pm_error:
pm_runtime_mark_last_busy(component->dev);
pm_runtime_put_autosuspend(component->dev);

14
sound/soc/sof/pm.c
View File

@ -164,7 +164,7 @@ static int sof_resume(struct device *dev, bool runtime_resume)
ret = tplg_ops->set_up_all_pipelines(sdev, false);
if (ret < 0) {
dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret);
return ret;
goto setup_fail;
}
}
@ -178,6 +178,18 @@ static int sof_resume(struct device *dev, bool runtime_resume)
dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret);
}
setup_fail:
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
if (ret < 0) {
/*
* Debugfs cannot be read in runtime suspend, so cache
* the contents upon failure. This allows to capture
* possible DSP coredump information.
*/
sof_cache_debugfs(sdev);
}
#endif
return ret;
}

14
sound/soc/sof/sof-client-probes.c
View File

@ -218,12 +218,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
ret = ipc->points_info(cdev, &desc, &num_desc);
if (ret < 0)
goto exit;
pm_runtime_mark_last_busy(dev);
err = pm_runtime_put_autosuspend(dev);
if (err < 0)
dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
goto pm_error;
for (i = 0; i < num_desc; i++) {
offset = strlen(buf);
@ -241,6 +236,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf));
kfree(desc);
pm_error:
pm_runtime_mark_last_busy(dev);
err = pm_runtime_put_autosuspend(dev);
if (err < 0)
dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
exit:
kfree(buf);
return ret;

6
sound/soc/sof/topology.c
View File

@ -586,6 +586,10 @@ static int sof_copy_tuples(struct snd_sof_dev *sdev, struct snd_soc_tplg_vendor_
if (*num_copied_tuples == tuples_size)
return 0;
}
/* stop when we've found the required token instances */
if (found == num_tokens * token_instance_num)
return 0;
}
/* next array */
@ -1261,7 +1265,7 @@ static int sof_widget_parse_tokens(struct snd_soc_component *scomp, struct snd_s
if (num_sets > 1) {
struct snd_sof_tuple *new_tuples;
num_tuples += token_list[object_token_list[i]].count * num_sets;
num_tuples += token_list[object_token_list[i]].count * (num_sets - 1);
new_tuples = krealloc(swidget->tuples,
sizeof(*new_tuples) * num_tuples, GFP_KERNEL);
if (!new_tuples) {