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authorLinus Torvalds <torvalds@linux-foundation.org>2020-03-07 07:59:30 -0600
committerLinus Torvalds <torvalds@linux-foundation.org>2020-03-07 07:59:30 -0600
commit676fc8de319335b840934c57ec75bdbbf049b738 (patch)
tree0496243bc783861fcd4e5f160980696f046b971a
parentMerge tag 'linux-kselftest-5.6-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/shuah/linux-kselftest (diff)
parentMerge tag 'asoc-fix-v5.6-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus (diff)
downloadwireguard-linux-676fc8de319335b840934c57ec75bdbbf049b738.tar.xz
wireguard-linux-676fc8de319335b840934c57ec75bdbbf049b738.zip
Merge tag 'sound-5.6-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai: "The regular "bump-in-the-middle" updates, containing mostly ASoC- related fixes at this time. All changes are reasonably small. A few entries are for ASoC and ALSA core parts (DAPM, PCM, topology) for followups of the recent changes and potential buffer overflow by snprintf(), while the rest are (both new and old) device-specific fixes for Intel, meson, tas2562, rt1015, as well as the usual HD-audio quirks" * tag 'sound-5.6-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits) ALSA: sgio2audio: Remove usage of dropped hw_params/hw_free functions ALSA: hda/realtek - Enable the headset of ASUS B9450FA with ALC294 ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1 ALSA: hda/realtek - Add Headset Mic supported ASoC: wm8741: Fix typo in Kconfig prompt ASoC: stm32: sai: manage rebind issue ASoC: SOF: Fix snd_sof_ipc_stream_posn() ASoC: rt1015: modify pre-divider for sysclk ASoC: rt1015: add operation callback function for rt1015_dai[] ASoC: soc-component: tidyup snd_soc_pcm_component_sync_stop() ASoC: dapm: Correct DAPM handling of active widgets during shutdown ASoC: tas2562: Fix sample rate error message ASoC: Intel: Skylake: Fix available clock counter incrementation ASoC: soc-pcm/soc-compress: don't use snd_soc_dapm_stream_stop() ASoC: meson: g12a: add tohdmitx reset ASoC: pcm512x: Fix unbalanced regulator enable call in probe error path ASoC: soc-core: fix for_rtd_codec_dai_rollback() macro ASoC: topology: Fix memleak in soc_tplg_manifest_load() ASoC: topology: Fix memleak in soc_tplg_link_elems_load() ...
-rw-r--r--include/sound/soc.h2
-rw-r--r--sound/mips/sgio2audio.c6
-rw-r--r--sound/pci/hda/patch_realtek.c31
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/pcm512x.c8
-rw-r--r--sound/soc/codecs/rt1015.c3
-rw-r--r--sound/soc/codecs/tas2562.c5
-rw-r--r--sound/soc/intel/skylake/skl-debug.c32
-rw-r--r--sound/soc/intel/skylake/skl-ssp-clk.c4
-rw-r--r--sound/soc/meson/g12a-tohdmitx.c6
-rw-r--r--sound/soc/soc-component.c2
-rw-r--r--sound/soc/soc-compress.c2
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/soc-pcm.c18
-rw-r--r--sound/soc/soc-topology.c17
-rw-r--r--sound/soc/sof/ipc.c2
-rw-r--r--sound/soc/stm/stm32_sai_sub.c18
17 files changed, 98 insertions, 62 deletions
diff --git a/include/sound/soc.h b/include/sound/soc.h
index f0e4f36f83bf..8a2266676b2d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1157,7 +1157,7 @@ struct snd_soc_pcm_runtime {
((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
(i)++)
#define for_each_rtd_codec_dai_rollback(rtd, i, dai) \
- for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);)
+ for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);)
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 9f60a5037f8b..5bf1ea150f26 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -649,8 +649,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
.open = snd_sgio2audio_playback1_open,
.close = snd_sgio2audio_pcm_close,
- .hw_params = snd_sgio2audio_pcm_hw_params,
- .hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
@@ -659,8 +657,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
.open = snd_sgio2audio_playback2_open,
.close = snd_sgio2audio_pcm_close,
- .hw_params = snd_sgio2audio_pcm_hw_params,
- .hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
@@ -669,8 +665,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
static const struct snd_pcm_ops snd_sgio2audio_capture_ops = {
.open = snd_sgio2audio_capture_open,
.close = snd_sgio2audio_pcm_close,
- .hw_params = snd_sgio2audio_pcm_hw_params,
- .hw_free = snd_sgio2audio_pcm_hw_free,
.prepare = snd_sgio2audio_pcm_prepare,
.trigger = snd_sgio2audio_pcm_trigger,
.pointer = snd_sgio2audio_pcm_pointer,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 477589e7ec1d..0ac06ff1a17c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
+ SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
@@ -5920,7 +5921,8 @@ enum {
ALC289_FIXUP_DUAL_SPK,
ALC294_FIXUP_SPK2_TO_DAC1,
ALC294_FIXUP_ASUS_DUAL_SPK,
-
+ ALC285_FIXUP_THINKPAD_HEADSET_JACK,
+ ALC294_FIXUP_ASUS_HPE,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -6684,6 +6686,8 @@ static const struct hda_fixup alc269_fixups[] = {
[ALC285_FIXUP_SPEAKER2_TO_DAC1] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
},
[ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = {
.type = HDA_FIXUP_PINS,
@@ -7040,7 +7044,23 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC294_FIXUP_SPK2_TO_DAC1
},
-
+ [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_jack,
+ .chained = true,
+ .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1
+ },
+ [ALC294_FIXUP_ASUS_HPE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Set EAPD high */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7115,6 +7135,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK),
SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK),
+ SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -7204,6 +7226,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
@@ -7274,8 +7297,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
- SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1),
- SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1),
+ SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7e90f5d83097..ea912439e446 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1406,7 +1406,7 @@ config SND_SOC_WM8737
depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8741
- tristate "Wolfson Microelectronics WM8737 DAC"
+ tristate "Wolfson Microelectronics WM8741 DAC"
depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8750
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 861210f6bf4f..4cbef9affffd 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1564,13 +1564,15 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap)
}
pcm512x->sclk = devm_clk_get(dev, NULL);
- if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
- return -EPROBE_DEFER;
+ if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) {
+ ret = -EPROBE_DEFER;
+ goto err;
+ }
if (!IS_ERR(pcm512x->sclk)) {
ret = clk_prepare_enable(pcm512x->sclk);
if (ret != 0) {
dev_err(dev, "Failed to enable SCLK: %d\n", ret);
- return ret;
+ goto err;
}
}
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 6d490e2dbc25..66eb55b4ffd4 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -664,7 +664,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream,
snd_soc_component_update_bits(component, RT1015_TDM_MASTER,
RT1015_I2S_DL_MASK, val_len);
snd_soc_component_update_bits(component, RT1015_CLK2,
- RT1015_FS_PD_MASK, pre_div);
+ RT1015_FS_PD_MASK, pre_div << RT1015_FS_PD_SFT);
return 0;
}
@@ -857,6 +857,7 @@ struct snd_soc_dai_driver rt1015_dai[] = {
.rates = RT1015_STEREO_RATES,
.formats = RT1015_FORMATS,
},
+ .ops = &rt1015_aif_dai_ops,
}
};
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 729acd874c48..be52886a5edb 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth)
break;
default:
- dev_info(tas2562->dev, "Not supported params format\n");
+ dev_info(tas2562->dev, "Unsupported bitwidth format\n");
+ return -EINVAL;
}
ret = snd_soc_component_update_bits(tas2562->component,
@@ -251,7 +252,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream,
ret = tas2562_set_samplerate(tas2562, params_rate(params));
if (ret)
- dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret);
+ dev_err(tas2562->dev, "set sample rate failed, %d\n", ret);
return ret;
}
diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index 3466675f2678..a15aa2ffa681 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -34,8 +34,8 @@ static ssize_t skl_print_pins(struct skl_module_pin *m_pin, char *buf,
int i;
ssize_t ret = 0;
- for (i = 0; i < max_pin; i++)
- ret += snprintf(buf + size, MOD_BUF - size,
+ for (i = 0; i < max_pin; i++) {
+ ret += scnprintf(buf + size, MOD_BUF - size,
"%s %d\n\tModule %d\n\tInstance %d\n\t"
"In-used %s\n\tType %s\n"
"\tState %d\n\tIndex %d\n",
@@ -45,13 +45,15 @@ static ssize_t skl_print_pins(struct skl_module_pin *m_pin, char *buf,
m_pin[i].in_use ? "Used" : "Unused",
m_pin[i].is_dynamic ? "Dynamic" : "Static",
m_pin[i].pin_state, i);
+ size += ret;
+ }
return ret;
}
static ssize_t skl_print_fmt(struct skl_module_fmt *fmt, char *buf,
ssize_t size, bool direction)
{
- return snprintf(buf + size, MOD_BUF - size,
+ return scnprintf(buf + size, MOD_BUF - size,
"%s\n\tCh %d\n\tFreq %d\n\tBit depth %d\n\t"
"Valid bit depth %d\n\tCh config %#x\n\tInterleaving %d\n\t"
"Sample Type %d\n\tCh Map %#x\n",
@@ -75,16 +77,16 @@ static ssize_t module_read(struct file *file, char __user *user_buf,
if (!buf)
return -ENOMEM;
- ret = snprintf(buf, MOD_BUF, "Module:\n\tUUID %pUL\n\tModule id %d\n"
+ ret = scnprintf(buf, MOD_BUF, "Module:\n\tUUID %pUL\n\tModule id %d\n"
"\tInstance id %d\n\tPvt_id %d\n", mconfig->guid,
mconfig->id.module_id, mconfig->id.instance_id,
mconfig->id.pvt_id);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Resources:\n\tCPC %#x\n\tIBS %#x\n\tOBS %#x\t\n",
res->cpc, res->ibs, res->obs);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Module data:\n\tCore %d\n\tIn queue %d\n\t"
"Out queue %d\n\tType %s\n",
mconfig->core_id, mconfig->max_in_queue,
@@ -94,38 +96,38 @@ static ssize_t module_read(struct file *file, char __user *user_buf,
ret += skl_print_fmt(mconfig->in_fmt, buf, ret, true);
ret += skl_print_fmt(mconfig->out_fmt, buf, ret, false);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Fixup:\n\tParams %#x\n\tConverter %#x\n",
mconfig->params_fixup, mconfig->converter);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Module Gateway:\n\tType %#x\n\tVbus %#x\n\tHW conn %#x\n\tSlot %#x\n",
mconfig->dev_type, mconfig->vbus_id,
mconfig->hw_conn_type, mconfig->time_slot);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Pipeline:\n\tID %d\n\tPriority %d\n\tConn Type %d\n\t"
"Pages %#x\n", mconfig->pipe->ppl_id,
mconfig->pipe->pipe_priority, mconfig->pipe->conn_type,
mconfig->pipe->memory_pages);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tParams:\n\t\tHost DMA %d\n\t\tLink DMA %d\n",
mconfig->pipe->p_params->host_dma_id,
mconfig->pipe->p_params->link_dma_id);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tPCM params:\n\t\tCh %d\n\t\tFreq %d\n\t\tFormat %d\n",
mconfig->pipe->p_params->ch,
mconfig->pipe->p_params->s_freq,
mconfig->pipe->p_params->s_fmt);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tLink %#x\n\tStream %#x\n",
mconfig->pipe->p_params->linktype,
mconfig->pipe->p_params->stream);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tState %d\n\tPassthru %s\n",
mconfig->pipe->state,
mconfig->pipe->passthru ? "true" : "false");
@@ -135,7 +137,7 @@ static ssize_t module_read(struct file *file, char __user *user_buf,
ret += skl_print_pins(mconfig->m_out_pin, buf,
mconfig->max_out_queue, ret, false);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Other:\n\tDomain %d\n\tHomogeneous Input %s\n\t"
"Homogeneous Output %s\n\tIn Queue Mask %d\n\t"
"Out Queue Mask %d\n\tDMA ID %d\n\tMem Pages %d\n\t"
@@ -191,7 +193,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf,
__ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
- ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
+ ret += scnprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
hex_dump_to_buffer(d->fw_read_buff + offset, 16, 16, 4,
tmp + ret, FW_REG_BUF - ret, 0);
ret += strlen(tmp + ret);
diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c
index 1c0e5226cb5b..bd43885f3805 100644
--- a/sound/soc/intel/skylake/skl-ssp-clk.c
+++ b/sound/soc/intel/skylake/skl-ssp-clk.c
@@ -384,9 +384,11 @@ static int skl_clk_dev_probe(struct platform_device *pdev)
&clks[i], clk_pdata, i);
if (IS_ERR(data->clk[data->avail_clk_cnt])) {
- ret = PTR_ERR(data->clk[data->avail_clk_cnt++]);
+ ret = PTR_ERR(data->clk[data->avail_clk_cnt]);
goto err_unreg_skl_clk;
}
+
+ data->avail_clk_cnt++;
}
platform_set_drvdata(pdev, data);
diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index 9cfbd343a00c..8a0db28a6a40 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <sound/pcm_params.h>
#include <linux/regmap.h>
+#include <linux/reset.h>
#include <sound/soc.h>
#include <sound/soc-dai.h>
@@ -378,6 +379,11 @@ static int g12a_tohdmitx_probe(struct platform_device *pdev)
struct device *dev = &pdev->dev;
void __iomem *regs;
struct regmap *map;
+ int ret;
+
+ ret = device_reset(dev);
+ if (ret)
+ return ret;
regs = devm_platform_ioremap_resource(pdev, 0);
if (IS_ERR(regs))
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 14e175cdeeb8..785a0385cc7f 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -451,7 +451,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream)
int i, ret;
for_each_rtd_components(rtd, i, component) {
- if (component->driver->ioctl) {
+ if (component->driver->sync_stop) {
ret = component->driver->sync_stop(component,
substream);
if (ret < 0)
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 223cd045719e..392a1c5b15d3 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -299,7 +299,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
for_each_dpcm_be(fe, stream, dpcm)
dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
- snd_soc_dapm_stream_stop(fe, stream);
+ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9b130561d562..9fb54e6fe254 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4772,7 +4772,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm)
continue;
if (w->power) {
dapm_seq_insert(w, &down_list, false);
- w->power = 0;
+ w->new_power = 0;
powerdown = 1;
}
}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ff1b7c7078e5..2c59b3688ca0 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2006,7 +2006,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
soc_pcm_close(substream);
/* run the stream event for each BE */
- snd_soc_dapm_stream_stop(fe, stream);
+ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
@@ -3171,16 +3171,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
unsigned long flags;
/* FE state */
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"[%s - %s]\n", fe->dai_link->name,
stream ? "Capture" : "Playback");
- offset += snprintf(buf + offset, size - offset, "State: %s\n",
+ offset += scnprintf(buf + offset, size - offset, "State: %s\n",
dpcm_state_string(fe->dpcm[stream].state));
if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
(fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"Hardware Params: "
"Format = %s, Channels = %d, Rate = %d\n",
snd_pcm_format_name(params_format(params)),
@@ -3188,10 +3188,10 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
params_rate(params));
/* BEs state */
- offset += snprintf(buf + offset, size - offset, "Backends:\n");
+ offset += scnprintf(buf + offset, size - offset, "Backends:\n");
if (list_empty(&fe->dpcm[stream].be_clients)) {
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" No active DSP links\n");
goto out;
}
@@ -3201,16 +3201,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be = dpcm->be;
params = &dpcm->hw_params;
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"- %s\n", be->dai_link->name);
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" State: %s\n",
dpcm_state_string(be->dpcm[stream].state));
if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" Hardware Params: "
"Format = %s, Channels = %d, Rate = %d\n",
snd_pcm_format_name(params_format(params)),
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index d2ee6ad20e83..575da6aba807 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -2377,8 +2377,11 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg,
}
ret = soc_tplg_link_config(tplg, _link);
- if (ret < 0)
+ if (ret < 0) {
+ if (!abi_match)
+ kfree(_link);
return ret;
+ }
/* offset by version-specific struct size and
* real priv data size
@@ -2542,7 +2545,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
{
struct snd_soc_tplg_manifest *manifest, *_manifest;
bool abi_match;
- int err;
+ int ret = 0;
if (tplg->pass != SOC_TPLG_PASS_MANIFEST)
return 0;
@@ -2555,19 +2558,19 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg,
_manifest = manifest;
} else {
abi_match = false;
- err = manifest_new_ver(tplg, manifest, &_manifest);
- if (err < 0)
- return err;
+ ret = manifest_new_ver(tplg, manifest, &_manifest);
+ if (ret < 0)
+ return ret;
}
/* pass control to component driver for optional further init */
if (tplg->comp && tplg->ops && tplg->ops->manifest)
- return tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
+ ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
if (!abi_match) /* free the duplicated one */
kfree(_manifest);
- return 0;
+ return ret;
}
/* validate header magic, size and type */
diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c
index b63fc529b456..78aa1da7c7a9 100644
--- a/sound/soc/sof/ipc.c
+++ b/sound/soc/sof/ipc.c
@@ -499,7 +499,7 @@ int snd_sof_ipc_stream_posn(struct snd_soc_component *scomp,
/* send IPC to the DSP */
err = sof_ipc_tx_message(sdev->ipc,
- stream.hdr.cmd, &stream, sizeof(stream), &posn,
+ stream.hdr.cmd, &stream, sizeof(stream), posn,
sizeof(*posn));
if (err < 0) {
dev_err(sdev->dev, "error: failed to get stream %d position\n",
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 30bcd5d3a32a..10eb4b8e8e7e 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
return ret;
}
- ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component,
- &sai->cpu_dai_drv, 1);
+ ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register pcm dma\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_component(&pdev->dev, &stm32_component,
+ &sai->cpu_dai_drv, 1);
if (ret)
return ret;
if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
conf = &stm32_sai_pcm_config_spdif;
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
- if (ret) {
- dev_err(&pdev->dev, "Could not register pcm dma\n");
- return ret;
- }
-
return 0;
}
@@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev)
struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev);
clk_unprepare(sai->pdata->pclk);
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}