diff --git a/include/sound/soc.h b/include/sound/soc.h
index c28a1ed5e8df9f4e82b783d0ac787606763c4258..26289679982647ca6feb176624c1db65948dfa76 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -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
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1fe581167b7b9aad19be6a06c3c5d98d3183cf45..d083225344a06d7a5aadb5b9afb5d7918f459c02 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -739,6 +739,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
runtime->boundary *= 2;
+ /* clear the buffer for avoiding possible kernel info leaks */
+ if (runtime->dma_area && !substream->ops->copy_user)
+ memset(runtime->dma_area, 0, runtime->dma_bytes);
+
snd_pcm_timer_resolution_change(substream);
snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index f9707fb05efe6812e9ec2ad69fb45e96b9bd9db5..682ed39f79b015a3982b6e149ff3bf1c0d3632b6 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev)
snd_hdac_stream_updateb(azx_dev, SD_CTL,
SD_CTL_DMA_START | SD_INT_MASK, 0);
snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
- if (azx_dev->stripe) {
+ if (azx_dev->stripe)
snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
- azx_dev->stripe = 0;
- }
azx_dev->running = false;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_clear);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2f3b7a35f2d9cdb3b9262d220c962336f6653205..ba56b59b3e17857747c533d2635e9b61e468a528 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -883,7 +883,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
return -EAGAIN; /* give a chance to retry */
}
- dev_WARN(chip->card->dev,
+ dev_err(chip->card->dev,
"azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n",
bus->last_cmd[addr]);
chip->single_cmd = 1;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7a1abb3e2318ecc1051e6b8bbef49ea6fa46f73..32ed46464af7ad0a3ec1197cb935db3a1571317f 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1809,13 +1809,14 @@ struct scp_msg {
static void dspio_clear_response_queue(struct hda_codec *codec)
{
+ unsigned long timeout = jiffies + msecs_to_jiffies(1000);
unsigned int dummy = 0;
- int status = -1;
+ int status;
/* clear all from the response queue */
do {
status = dspio_read(codec, &dummy);
- } while (status == 0);
+ } while (status == 0 && time_before(jiffies, timeout));
}
static int dspio_get_response_data(struct hda_codec *codec)
@@ -7588,12 +7589,14 @@ static void ca0132_process_dsp_response(struct hda_codec *codec,
struct ca0132_spec *spec = codec->spec;
codec_dbg(codec, "ca0132_process_dsp_response\n");
+ snd_hda_power_up_pm(codec);
if (spec->wait_scp) {
if (dspio_get_response_data(codec) >= 0)
spec->wait_scp = 0;
}
dspio_clear_response_queue(codec);
+ snd_hda_power_down_pm(codec);
}
static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -7604,11 +7607,10 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
/* Delay enabling the HP amp, to let the mic-detection
* state machine run.
*/
- cancel_delayed_work(&spec->unsol_hp_work);
- schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
tbl = snd_hda_jack_tbl_get(codec, cb->nid);
if (tbl)
tbl->block_report = 1;
+ schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
}
static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -8454,12 +8456,25 @@ static void ca0132_reboot_notify(struct hda_codec *codec)
codec->patch_ops.free(codec);
}
+#ifdef CONFIG_PM
+static int ca0132_suspend(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ cancel_delayed_work_sync(&spec->unsol_hp_work);
+ return 0;
+}
+#endif
+
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
+#ifdef CONFIG_PM
+ .suspend = ca0132_suspend,
+#endif
.reboot_notify = ca0132_reboot_notify,
};
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78647ee02339fbe36d1374dbf7fbbec946b458f0..630b1f5c276d4eb348e65603ffbcd1e6e087660b 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
per_cvt->assigned = 0;
hinfo->nid = 0;
+ azx_stream(get_azx_dev(substream))->stripe = 0;
+
mutex_lock(&spec->pcm_lock);
snd_hda_spdif_ctls_unassign(codec, pcm_idx);
clear_bit(pcm_idx, &spec->pcm_in_use);
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index f4ee6798154af52377c7a6949e8fc99a2b0134fe..7a5621e5e2330dd872e3835811862c9b339dbfd8 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -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,
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f6bf4cfbea23fa0aae7e5af9ec933870597fb278..e46b6ada13b1b01af12a8a775cc4fcc8d6e95ef6 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -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;
}
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 57965cd678b4e5ae7de6865dacd7ff4c8eee793a..a197114b0dad3fc648952b8f272fb8b87c1a77c1 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -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;
diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h
index 3af36ec928e93480ba4332430364917d926f5bd7..088b779317276f266ed4cd59571dbb8245a6d35f 100644
--- a/sound/soc/codecs/rt5677-spi.h
+++ b/sound/soc/codecs/rt5677-spi.h
@@ -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__ */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index b1713fffa3eb925f3d79f6d23b2c2bbb2bd38ddd..ae6f6121bc1b0ffadef6f57828b46b0fe89da958 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -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);
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 7d7ea15d73e0a677c61b094fe1e91b592378a7f7..5ffbaddd6e499579ef98d3c35b146ab645fc2c4b 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -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) {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3e5c69fbc33ad752afa07868d0df00444aeb920d..d9d59f45833fdd2f02a3f202b80941535e39f1ca 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -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 */
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 10b82bf043d1ab190e50895c67907c30b8cf6457..55e9f8800b3e103638e38b651aa9e37ac37fdcc7 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -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,
diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c
index fbecbb74350b51de1583b2be7bec5408fef46b7b..68bcec5241f74fb6da8345f722b1ed7a43f6decf 100644
--- a/sound/soc/intel/atom/sst/sst.c
+++ b/sound/soc/intel/atom/sst/sst.c
@@ -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>
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index dd2b5ad08659107f41184234086eb81b10608554..243f683bc02a7361fb89ff2eaa7ab1f6e1ab751c 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -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 = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index 5d08ae0667380bea84b98537cb4d2c66c3a76281..fb9ba881970604331a6d825f09893609eae9ada2 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -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,
+ .id = "10EC5682",
+ .drv_name = "cml_rt1011_rt5682",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &rt1011_spk_codecs,
.sof_fw_filename = "sof-cml.ri",
- .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
+ .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
},
{
- .id = "MX98357A",
+ .id = "10EC5682",
.drv_name = "sof_rt5682",
- .quirk_data = &cml_codecs,
+ .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 = "10EC1011",
- .drv_name = "cml_rt1011_rt5682",
- .quirk_data = &cml_codecs,
- .sof_fw_filename = "sof-cml.ri",
- .sof_tplg_filename = "sof-cml-rt1011-rt5682.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);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 61f230324164d0a6765aaba8b55df82982b38263..6615ef64c7f55ad1c7bd94f186babaa64feab966 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -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;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 062653ab03a37b1cac41e11f1a5984805b8ca688..1c84ff1a5bf96b09e303ec08d5d4c6e1e565aad9 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -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
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 76b7ee637e869780dce61b0f887309692a08849d..01e7bc03d92f96e9e6b9fc0a16453cd48fdfb8b7 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -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;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 81d2af000a5c80b2e7f7c36a31029414d32d95c3..b28613149b0cce2e4a2b1623091fac095808818b 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -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
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 2abf80b3eb52fee2c0e7f06c9c6e051028f2104d..92ef6a796fd5ec8dd554a53d76d9da306c48792e 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -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);
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 9a9a381a908dee467665c979fecc7d31fb58e3bc..432d12bd493791837ff38f8afe0e71c35d858915 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -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;
}
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index d82ab981e8408ad0f9ea0d9179eff846976a7284..e20b806ec80f1aa1cb74989baca3130c8ff04427 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -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);