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ASoC: Fixes for v5.5

Browse Source 
A collection of fixes since the merge window, mostly driver specific but
 there's a few in the core that clean up fallout from the refactorings
 done in the last cycle.
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Merge tag 'asoc-fix-v5.5-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.5

A collection of fixes since the merge window, mostly driver specific but
there's a few in the core that clean up fallout from the refactorings
done in the last cycle.
This commit is contained in:
Takashi Iwai 2019-12-17 14:18:32 +01:00
commit 7c497d7992
19 changed files with 161 additions and 100 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
include/sound/soc.h
View File

@ -1150,6 +1150,7 @@ struct snd_soc_pcm_runtime {
unsigned int num_codecs;
struct delayed_work delayed_work;
void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dpcm_root;
#endif

46
sound/soc/amd/acp-da7219-max98357a.c
View File

@ -96,14 +96,19 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
static int da7219_clk_enable(struct snd_pcm_substream *substream,
int wclk_rate, int bclk_rate)
static int da7219_clk_enable(struct snd_pcm_substream *substream)
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
clk_set_rate(da7219_dai_wclk, wclk_rate);
clk_set_rate(da7219_dai_bclk, bclk_rate);
/*
* Set wclk to 48000 because the rate constraint of this driver is
* 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is
* minimum of 64x the LRCLK sample rate." DA7219 is the only clk
* source so for all codecs we have to limit bclk to 64X lrclk.
*/
clk_set_rate(da7219_dai_wclk, 48000);
clk_set_rate(da7219_dai_bclk, 48000 * 64);
ret = clk_prepare_enable(da7219_dai_bclk);
if (ret < 0) {
dev_err(rtd->dev, "can't enable master clock %d\n", ret);
@ -156,7 +161,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
&constraints_rates);
machine->play_i2s_instance = I2S_SP_INSTANCE;
return 0;
return da7219_clk_enable(substream);
}
static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
@ -178,7 +183,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
machine->cap_i2s_instance = I2S_SP_INSTANCE;
machine->capture_channel = CAP_CHANNEL1;
return 0;
return da7219_clk_enable(substream);
}
static int cz_max_startup(struct snd_pcm_substream *substream)
@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
&constraints_rates);
machine->play_i2s_instance = I2S_BT_INSTANCE;
return 0;
return da7219_clk_enable(substream);
}
static int cz_dmic0_startup(struct snd_pcm_substream *substream)
@ -220,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream)
&constraints_rates);
machine->cap_i2s_instance = I2S_BT_INSTANCE;
return 0;
return da7219_clk_enable(substream);
}
static int cz_dmic1_startup(struct snd_pcm_substream *substream)
@ -242,25 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream)
machine->cap_i2s_instance = I2S_SP_INSTANCE;
machine->capture_channel = CAP_CHANNEL0;
return 0;
}
static int cz_da7219_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
int wclk, bclk;
wclk = params_rate(params);
bclk = wclk * params_channels(params) *
snd_pcm_format_width(params_format(params));
/* ADAU7002 spec: "The ADAU7002 requires a BCLK rate
* that is minimum of 64x the LRCLK sample rate."
* DA7219 is the only clk source so for all codecs
* we have to limit bclk to 64X lrclk.
*/
if (bclk < (wclk * 64))
bclk = wclk * 64;
return da7219_clk_enable(substream, wclk, bclk);
return da7219_clk_enable(substream);
}
static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
@ -271,31 +258,26 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
static const struct snd_soc_ops cz_da7219_play_ops = {
.startup = cz_da7219_play_startup,
.shutdown = cz_da7219_shutdown,
.hw_params = cz_da7219_params,
};
static const struct snd_soc_ops cz_da7219_cap_ops = {
.startup = cz_da7219_cap_startup,
.shutdown = cz_da7219_shutdown,
.hw_params = cz_da7219_params,
};
static const struct snd_soc_ops cz_max_play_ops = {
.startup = cz_max_startup,
.shutdown = cz_da7219_shutdown,
.hw_params = cz_da7219_params,
};
static const struct snd_soc_ops cz_dmic0_cap_ops = {
.startup = cz_dmic0_startup,
.shutdown = cz_da7219_shutdown,
.hw_params = cz_da7219_params,
};
static const struct snd_soc_ops cz_dmic1_cap_ops = {
.startup = cz_dmic1_startup,
.shutdown = cz_da7219_shutdown,
.hw_params = cz_da7219_params,
};
SND_SOC_DAILINK_DEF(designware1,

30
sound/soc/codecs/max98090.c
View File

@ -2103,26 +2103,40 @@ static void max98090_pll_det_disable_work(struct work_struct *work)
M98090_IULK_MASK, 0);
}
static void max98090_pll_work(struct work_struct *work)
static void max98090_pll_work(struct max98090_priv *max98090)
{
struct max98090_priv *max98090 =
container_of(work, struct max98090_priv, pll_work);
struct snd_soc_component *component = max98090->component;
unsigned int pll;
int i;
if (!snd_soc_component_is_active(component))
return;
dev_info_ratelimited(component->dev, "PLL unlocked\n");
/*
* As the datasheet suggested, the maximum PLL lock time should be
* 7 msec. The workaround resets the codec softly by toggling SHDN
* off and on if PLL failed to lock for 10 msec. Notably, there is
* no suggested hold time for SHDN off.
*/
/* Toggle shutdown OFF then ON */
snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
M98090_SHDNN_MASK, 0);
msleep(10);
snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
M98090_SHDNN_MASK, M98090_SHDNN_MASK);
/* Give PLL time to lock */
msleep(10);
for (i = 0; i < 10; ++i) {
/* Give PLL time to lock */
usleep_range(1000, 1200);
/* Check lock status */
pll = snd_soc_component_read32(
component, M98090_REG_DEVICE_STATUS);
if (!(pll & M98090_ULK_MASK))
break;
}
}
static void max98090_jack_work(struct work_struct *work)
@ -2259,7 +2273,7 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
if (active & M98090_ULK_MASK) {
dev_dbg(component->dev, "M98090_ULK_MASK\n");
schedule_work(&max98090->pll_work);
max98090_pll_work(max98090);
}
if (active & M98090_JDET_MASK) {
@ -2422,7 +2436,6 @@ static int max98090_probe(struct snd_soc_component *component)
max98090_pll_det_enable_work);
INIT_WORK(&max98090->pll_det_disable_work,
max98090_pll_det_disable_work);
INIT_WORK(&max98090->pll_work, max98090_pll_work);
/* Enable jack detection */
snd_soc_component_write(component, M98090_REG_JACK_DETECT,
@ -2475,7 +2488,6 @@ static void max98090_remove(struct snd_soc_component *component)
cancel_delayed_work_sync(&max98090->jack_work);
cancel_delayed_work_sync(&max98090->pll_det_enable_work);
cancel_work_sync(&max98090->pll_det_disable_work);
cancel_work_sync(&max98090->pll_work);
max98090->component = NULL;
}

1
sound/soc/codecs/max98090.h
View File

@ -1530,7 +1530,6 @@ struct max98090_priv {
struct delayed_work jack_work;
struct delayed_work pll_det_enable_work;
struct work_struct pll_det_disable_work;
struct work_struct pll_work;
struct snd_soc_jack *jack;
unsigned int dai_fmt;
int tdm_slots;

16
sound/soc/codecs/rt5677-spi.h
View File

@ -9,9 +9,25 @@
#ifndef __RT5677_SPI_H__
#define __RT5677_SPI_H__
#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)
int rt5677_spi_read(u32 addr, void *rxbuf, size_t len);
int rt5677_spi_write(u32 addr, const void *txbuf, size_t len);
int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw);
void rt5677_spi_hotword_detected(void);
#else
static inline int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
{
return -EINVAL;
}
static inline int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
{
return -EINVAL;
}
static inline int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw)
{
return -EINVAL;
}
static inline void rt5677_spi_hotword_detected(void){}
#endif
#endif /* __RT5677_SPI_H__ */

2
sound/soc/codecs/rt5682.c
View File

@ -73,6 +73,7 @@ struct rt5682_priv {
static const struct reg_sequence patch_list[] = {
{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
{RT5682_I2C_CTRL, 0x000f},
};
static const struct reg_default rt5682_reg[] = {
@ -2474,6 +2475,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
mutex_lock(&rt5682->calibrate_mutex);
rt5682_reset(rt5682->regmap);
regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f);
regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
usleep_range(15000, 20000);
regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);

6
sound/soc/codecs/wm8904.c
View File

@ -1806,6 +1806,12 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id,
switch (clk_id) {
case WM8904_CLK_AUTO:
/* We don't have any rate constraints, so just ignore the
* request to disable constraining.
*/
if (!freq)
return 0;
mclk_freq = clk_get_rate(priv->mclk);
/* enable FLL if a different sysclk is desired */
if (mclk_freq != freq) {

4
sound/soc/codecs/wm8962.c
View File

@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
if (target % Fref == 0) {
fll_div->theta = 0;
fll_div->lambda = 0;
fll_div->lambda = 1;
} else {
gcd_fll = gcd(target, fratio * Fref);
@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s
return -EINVAL;
}
if (fll_div.theta || fll_div.lambda)
if (fll_div.theta)
fll1 |= WM8962_FLL_FRAC;
/* Stop the FLL while we reconfigure */

6
sound/soc/generic/simple-card.c
View File

@ -371,6 +371,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
do {
struct asoc_simple_data adata;
struct device_node *codec;
struct device_node *plat;
struct device_node *np;
int num = of_get_child_count(node);
@ -381,6 +382,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
ret = -ENODEV;
goto error;
}
/* get platform */
plat = of_get_child_by_name(node, is_top ?
PREFIX "plat" : "plat");
/* get convert-xxx property */
memset(&adata, 0, sizeof(adata));
@ -389,6 +393,8 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
/* loop for all CPU/Codec node */
for_each_child_of_node(node, np) {
if (plat == np)
continue;
/*
* It is DPCM
* if it has many CPUs,

1
sound/soc/intel/atom/sst/sst.c
View File

@ -14,6 +14,7 @@
#include <linux/module.h>
#include <linux/fs.h>
#include <linux/interrupt.h>
#include <linux/io.h>
#include <linux/firmware.h>
#include <linux/pm_runtime.h>
#include <linux/pm_qos.h>

8
sound/soc/intel/boards/bytcr_rt5640.c
View File

@ -707,13 +707,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
BYT_RT5640_MCLK_EN),
},
{
/* Teclast X89 */
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"),
DMI_MATCH(DMI_BOARD_NAME, "tPAD"),
},
.driver_data = (void *)(BYT_RT5640_IN3_MAP |
BYT_RT5640_MCLK_EN |
BYT_RT5640_SSP0_AIF1),
BYT_RT5640_JD_SRC_JD1_IN4P |
BYT_RT5640_OVCD_TH_2000UA |
BYT_RT5640_OVCD_SF_1P0 |
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
{ /* Toshiba Satellite Click Mini L9W-B */
.matches = {

47
sound/soc/intel/common/soc-acpi-intel-cml-match.c
View File

@ -9,45 +9,52 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
static struct snd_soc_acpi_codecs cml_codecs = {
static struct snd_soc_acpi_codecs rt1011_spk_codecs = {
.num_codecs = 1,
.codecs = {"10EC5682"}
.codecs = {"10EC1011"}
};
static struct snd_soc_acpi_codecs cml_spk_codecs = {
static struct snd_soc_acpi_codecs max98357a_spk_codecs = {
.num_codecs = 1,
.codecs = {"MX98357A"}
};
/*
* The order of the three entries with .id = "10EC5682" matters
* here, because DSDT tables expose an ACPI HID for the MAX98357A
* speaker amplifier which is not populated on the board.
*/
struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
{
.id = "DLGS7219",
.drv_name = "cml_da7219_max98357a",
.quirk_data = &cml_spk_codecs,
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
},
{
.id = "MX98357A",
.drv_name = "sof_rt5682",
.quirk_data = &cml_codecs,
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg",
},
{
.id = "10EC1011",
.id = "10EC5682",
.drv_name = "cml_rt1011_rt5682",
.quirk_data = &cml_codecs,
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &rt1011_spk_codecs,
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
},
{
.id = "10EC5682",
.drv_name = "sof_rt5682",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &max98357a_spk_codecs,
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg",
},
{
.id = "10EC5682",
.drv_name = "sof_rt5682",
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-rt5682.tplg",
},
{
.id = "DLGS7219",
.drv_name = "cml_da7219_max98357a",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &max98357a_spk_codecs,
.sof_fw_filename = "sof-cml.ri",
.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
},
{},
};
EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines);

6
sound/soc/soc-compress.c
View File

@ -214,10 +214,8 @@ be_err:
* This is to ensure there are no pops or clicks in between any music tracks
* due to DAPM power cycling.
*/
static void close_delayed_work(struct work_struct *work)
static void close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_pcm_runtime *rtd =
container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
struct snd_soc_dai *codec_dai = rtd->codec_dai;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@ -929,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
}
/* DAPM dai link stream work */
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
rtd->close_delayed_work_func = close_delayed_work;
rtd->compr = compr;
compr->private_data = rtd;

13
sound/soc/soc-core.c
View File

@ -419,7 +419,8 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
list_del(&rtd->list);
flush_delayed_work(&rtd->delayed_work);
if (delayed_work_pending(&rtd->delayed_work))
flush_delayed_work(&rtd->delayed_work);
snd_soc_pcm_component_free(rtd);
/*
@ -435,6 +436,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
device_unregister(rtd->dev);
}
static void close_delayed_work(struct work_struct *work) {
struct snd_soc_pcm_runtime *rtd =
container_of(work, struct snd_soc_pcm_runtime,
delayed_work.work);
if (rtd->close_delayed_work_func)
rtd->close_delayed_work_func(rtd);
}
static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
struct snd_soc_card *card, struct snd_soc_dai_link *dai_link)
{
@ -470,6 +480,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
rtd->dev = dev;
dev_set_drvdata(dev, rtd);
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
/*
* for rtd->codec_dais

11
sound/soc/soc-pcm.c
View File

@ -637,10 +637,8 @@ out:
* This is to ensure there are no pops or clicks in between any music tracks
* due to DAPM power cycling.
*/
static void close_delayed_work(struct work_struct *work)
static void close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_pcm_runtime *rtd =
container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
struct snd_soc_dai *codec_dai = rtd->codec_dais[0];
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work)
mutex_unlock(&rtd->card->pcm_mutex);
}
static void codec2codec_close_delayed_work(struct work_struct *work)
static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
/*
* Currently nothing to do for c2c links
@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
/* DAPM dai link stream work */
if (rtd->dai_link->params)
INIT_DELAYED_WORK(&rtd->delayed_work,
codec2codec_close_delayed_work);
rtd->close_delayed_work_func = codec2codec_close_delayed_work;
else
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
rtd->close_delayed_work_func = close_delayed_work;
pcm->nonatomic = rtd->dai_link->nonatomic;
rtd->pcm = pcm;

27
sound/soc/soc-topology.c
View File

@ -1933,11 +1933,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
ret = soc_tplg_dai_link_load(tplg, link, NULL);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
kfree(link->name);
kfree(link->stream_name);
kfree(link->cpus->dai_name);
kfree(link);
return ret;
goto err;
}
ret = snd_soc_add_dai_link(tplg->comp->card, link);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: adding FE link failed\n");
goto err;
}
link->dobj.index = tplg->index;
@ -1945,8 +1947,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
link->dobj.type = SND_SOC_DOBJ_DAI_LINK;
list_add(&link->dobj.list, &tplg->comp->dobj_list);
snd_soc_add_dai_link(tplg->comp->card, link);
return 0;
err:
kfree(link->name);
kfree(link->stream_name);
kfree(link->cpus->dai_name);
kfree(link);
return ret;
}
/* create a FE DAI and DAI link from the PCM object */
@ -2039,6 +2046,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
int size;
int i;
bool abi_match;
int ret;
count = le32_to_cpu(hdr->count);
@ -2080,7 +2088,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
}
/* create the FE DAIs and DAI links */
soc_tplg_pcm_create(tplg, _pcm);
ret = soc_tplg_pcm_create(tplg, _pcm);
if (ret < 0) {
if (!abi_match)
kfree(_pcm);
return ret;
}
/* offset by version-specific struct size and
* real priv data size

25
sound/soc/sof/intel/byt.c
View File

@ -24,7 +24,8 @@
#define DRAM_OFFSET 0x100000
#define DRAM_SIZE (160 * 1024)
#define SHIM_OFFSET 0x140000
#define SHIM_SIZE 0x100
#define SHIM_SIZE_BYT 0x100
#define SHIM_SIZE_CHT 0x118
#define MBOX_OFFSET 0x144000
#define MBOX_SIZE 0x1000
#define EXCEPT_OFFSET 0x800
@ -75,7 +76,7 @@ static const struct snd_sof_debugfs_map byt_debugfs[] = {
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_BYT,
SOF_DEBUGFS_ACCESS_ALWAYS},
};
@ -102,7 +103,7 @@ static const struct snd_sof_debugfs_map cht_debugfs[] = {
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_CHT,
SOF_DEBUGFS_ACCESS_ALWAYS},
};
@ -145,33 +146,33 @@ static void byt_dump(struct snd_sof_dev *sdev, u32 flags)
struct sof_ipc_dsp_oops_xtensa xoops;
struct sof_ipc_panic_info panic_info;
u32 stack[BYT_STACK_DUMP_SIZE];
u32 status, panic, imrd, imrx;
u64 status, panic, imrd, imrx;
/* now try generic SOF status messages */
status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD);
panic = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCX);
status = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCD);
panic = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCX);
byt_get_registers(sdev, &xoops, &panic_info, stack,
BYT_STACK_DUMP_SIZE);
snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack,
BYT_STACK_DUMP_SIZE);
/* provide some context for firmware debug */
imrx = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRX);
imrd = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRD);
imrx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRX);
imrd = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRD);
dev_err(sdev->dev,
"error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n",
"error: ipc host -> DSP: pending %s complete %s raw 0x%llx\n",
(panic & SHIM_IPCX_BUSY) ? "yes" : "no",
(panic & SHIM_IPCX_DONE) ? "yes" : "no", panic);
dev_err(sdev->dev,
"error: mask host: pending %s complete %s raw 0x%8.8x\n",
"error: mask host: pending %s complete %s raw 0x%llx\n",
(imrx & SHIM_IMRX_BUSY) ? "yes" : "no",
(imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx);
dev_err(sdev->dev,
"error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n",
"error: ipc DSP -> host: pending %s complete %s raw 0x%llx\n",
(status & SHIM_IPCD_BUSY) ? "yes" : "no",
(status & SHIM_IPCD_DONE) ? "yes" : "no", status);
dev_err(sdev->dev,
"error: mask DSP: pending %s complete %s raw 0x%8.8x\n",
"error: mask DSP: pending %s complete %s raw 0x%llx\n",
(imrd & SHIM_IMRD_BUSY) ? "yes" : "no",
(imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd);

7
sound/soc/sof/loader.c
View File

@ -50,8 +50,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
while (ext_hdr->hdr.cmd == SOF_IPC_FW_READY) {
/* read in ext structure */
offset += sizeof(*ext_hdr);
snd_sof_dsp_block_read(sdev, bar, offset,
snd_sof_dsp_block_read(sdev, bar, offset + sizeof(*ext_hdr),
(void *)((u8 *)ext_data + sizeof(*ext_hdr)),
ext_hdr->hdr.size - sizeof(*ext_hdr));
@ -61,11 +60,15 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
/* process structure data */
switch (ext_hdr->type) {
case SOF_IPC_EXT_DMA_BUFFER:
ret = 0;
break;
case SOF_IPC_EXT_WINDOW:
ret = get_ext_windows(sdev, ext_hdr);
break;
default:
dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
ext_hdr->type, ext_hdr->hdr.size);
ret = 0;
break;
}

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

@ -3132,7 +3132,9 @@ found:
case SOF_DAI_INTEL_SSP:
case SOF_DAI_INTEL_DMIC:
case SOF_DAI_INTEL_ALH:
/* no resource needs to be released for SSP, DMIC and ALH */
case SOF_DAI_IMX_SAI:
case SOF_DAI_IMX_ESAI:
/* no resource needs to be released for all cases above */
break;
case SOF_DAI_INTEL_HDA:
ret = sof_link_hda_unload(sdev, link);