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-rw-r--r--sound/core/pcm_native.c4
-rw-r--r--sound/firewire/fireface/ff-pcm.c2
-rw-r--r--sound/firewire/motu/motu-pcm.c8
-rw-r--r--sound/firewire/oxfw/oxfw-pcm.c2
-rw-r--r--sound/hda/hdac_stream.c4
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c20
-rw-r--r--sound/pci/hda/hda_controller.c2
-rw-r--r--sound/pci/hda/hda_intel.c22
-rw-r--r--sound/pci/hda/patch_ca0132.c23
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c52
-rw-r--r--sound/pci/ice1712/ice1724.c9
-rw-r--r--sound/soc/amd/acp-da7219-max98357a.c46
-rw-r--r--sound/soc/codecs/hdmi-codec.c2
-rw-r--r--sound/soc/codecs/max98090.c30
-rw-r--r--sound/soc/codecs/max98090.h1
-rw-r--r--sound/soc/codecs/rt5677-spi.h16
-rw-r--r--sound/soc/codecs/rt5682.c2
-rw-r--r--sound/soc/codecs/wm8904.c6
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/generic/simple-card.c6
-rw-r--r--sound/soc/intel/atom/sst/sst.c1
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c8
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cml-match.c41
-rw-r--r--sound/soc/soc-compress.c6
-rw-r--r--sound/soc/soc-core.c13
-rw-r--r--sound/soc/soc-pcm.c11
-rw-r--r--sound/soc/soc-topology.c27
-rw-r--r--sound/soc/sof/intel/byt.c25
-rw-r--r--sound/soc/sof/loader.c7
-rw-r--r--sound/soc/sof/topology.c4
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/pcm.c25
-rw-r--r--sound/usb/quirks-table.h3
-rw-r--r--sound/usb/quirks.c11
-rw-r--r--sound/usb/usbaudio.h3
36 files changed, 289 insertions, 160 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1fe581167b7b..d083225344a0 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/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index 4e3bd9a2bec0..bd91c6ecb112 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -247,7 +247,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&ff->mutex);
}
- return 0;
+ return err;
}
static int pcm_hw_free(struct snd_pcm_substream *substream)
diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c
index 349b4d09e84f..005970931030 100644
--- a/sound/firewire/motu/motu-pcm.c
+++ b/sound/firewire/motu/motu-pcm.c
@@ -177,18 +177,14 @@ static int pcm_open(struct snd_pcm_substream *substream)
err = snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
frames_per_period, frames_per_period);
- if (err < 0) {
- mutex_unlock(&motu->mutex);
+ if (err < 0)
goto err_locked;
- }
err = snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
frames_per_buffer, frames_per_buffer);
- if (err < 0) {
- mutex_unlock(&motu->mutex);
+ if (err < 0)
goto err_locked;
- }
}
}
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9124603edabe..67fd3e844dd6 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -285,7 +285,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&oxfw->mutex);
}
- return 0;
+ return err;
}
static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index f9707fb05efe..682ed39f79b0 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/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 50d4a87a6bb3..f02f5b1568de 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -635,36 +635,30 @@ This function assumes there are no more than 16 in/out busses or pipes
Meters is an array [3][16][2] of long. */
static void get_audio_meters(struct echoaudio *chip, long *meters)
{
- int i, m, n;
+ unsigned int i, m, n;
- m = 0;
- n = 0;
- for (i = 0; i < num_busses_out(chip); i++, m++) {
+ for (i = 0 ; i < 96; i++)
+ meters[i] = 0;
+
+ for (m = 0, n = 0, i = 0; i < num_busses_out(chip); i++, m++) {
meters[n++] = chip->comm_page->vu_meter[m];
meters[n++] = chip->comm_page->peak_meter[m];
}
- for (; n < 32; n++)
- meters[n] = 0;
#ifdef ECHOCARD_ECHO3G
m = E3G_MAX_OUTPUTS; /* Skip unused meters */
#endif
- for (i = 0; i < num_busses_in(chip); i++, m++) {
+ for (n = 32, i = 0; i < num_busses_in(chip); i++, m++) {
meters[n++] = chip->comm_page->vu_meter[m];
meters[n++] = chip->comm_page->peak_meter[m];
}
- for (; n < 64; n++)
- meters[n] = 0;
-
#ifdef ECHOCARD_HAS_VMIXER
- for (i = 0; i < num_pipes_out(chip); i++, m++) {
+ for (n = 64, i = 0; i < num_pipes_out(chip); i++, m++) {
meters[n++] = chip->comm_page->vu_meter[m];
meters[n++] = chip->comm_page->peak_meter[m];
}
#endif
- for (; n < 96; n++)
- meters[n] = 0;
}
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2f3b7a35f2d9..ba56b59b3e17 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/hda_intel.c b/sound/pci/hda/hda_intel.c
index 35b4526f0d28..5b92f290cbb0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -282,12 +282,13 @@ enum {
/* quirks for old Intel chipsets */
#define AZX_DCAPS_INTEL_ICH \
- (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE)
+ (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\
+ AZX_DCAPS_SYNC_WRITE)
/* quirks for Intel PCH */
#define AZX_DCAPS_INTEL_PCH_BASE \
(AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
/* PCH up to IVB; no runtime PM; bind with i915 gfx */
#define AZX_DCAPS_INTEL_PCH_NOPM \
@@ -302,13 +303,13 @@ enum {
#define AZX_DCAPS_INTEL_HASWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
/* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
#define AZX_DCAPS_INTEL_BROADWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
#define AZX_DCAPS_INTEL_BAYTRAIL \
(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT)
@@ -1410,7 +1411,17 @@ static bool atpx_present(void)
acpi_handle dhandle, atpx_handle;
acpi_status status;
- while ((pdev = pci_get_class(PCI_BASE_CLASS_DISPLAY << 16, pdev)) != NULL) {
+ while ((pdev = pci_get_class(PCI_CLASS_DISPLAY_VGA << 8, pdev)) != NULL) {
+ dhandle = ACPI_HANDLE(&pdev->dev);
+ if (dhandle) {
+ status = acpi_get_handle(dhandle, "ATPX", &atpx_handle);
+ if (!ACPI_FAILURE(status)) {
+ pci_dev_put(pdev);
+ return true;
+ }
+ }
+ }
+ while ((pdev = pci_get_class(PCI_CLASS_DISPLAY_OTHER << 8, pdev)) != NULL) {
dhandle = ACPI_HANDLE(&pdev->dev);
if (dhandle) {
status = acpi_get_handle(dhandle, "ATPX", &atpx_handle);
@@ -1419,7 +1430,6 @@ static bool atpx_present(void)
return true;
}
}
- pci_dev_put(pdev);
}
return false;
}
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7a1abb3e231..32ed46464af7 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 78647ee02339..630b1f5c276d 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/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d6e34b3b3aa..1cd4906a67e1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -501,6 +501,7 @@ static void alc_shutup_pins(struct hda_codec *codec)
struct alc_spec *spec = codec->spec;
switch (codec->core.vendor_id) {
+ case 0x10ec0283:
case 0x10ec0286:
case 0x10ec0288:
case 0x10ec0298:
@@ -5904,9 +5905,12 @@ enum {
ALC256_FIXUP_ASUS_HEADSET_MIC,
ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
ALC299_FIXUP_PREDATOR_SPK,
- ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
- ALC294_FIXUP_ASUS_INTSPK_GPIO,
+ ALC289_FIXUP_DELL_SPK2,
+ ALC289_FIXUP_DUAL_SPK,
+ ALC294_FIXUP_SPK2_TO_DAC1,
+ ALC294_FIXUP_ASUS_DUAL_SPK,
+
};
static const struct hda_fixup alc269_fixups[] = {
@@ -6981,33 +6985,45 @@ static const struct hda_fixup alc269_fixups[] = {
{ }
}
},
- [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = {
+ [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- { 0x14, 0x411111f0 }, /* disable confusing internal speaker */
- { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */
+ { 0x19, 0x04a11040 },
+ { 0x21, 0x04211020 },
{ }
},
.chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
},
- [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
+ [ALC289_FIXUP_DELL_SPK2] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
- { 0x19, 0x04a11040 },
- { 0x21, 0x04211020 },
+ { 0x17, 0x90170130 }, /* bass spk */
{ }
},
.chained = true,
- .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+ .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE
},
- [ALC294_FIXUP_ASUS_INTSPK_GPIO] = {
+ [ALC289_FIXUP_DUAL_SPK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC289_FIXUP_DELL_SPK2
+ },
+ [ALC294_FIXUP_SPK2_TO_DAC1] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+ },
+ [ALC294_FIXUP_ASUS_DUAL_SPK] = {
.type = HDA_FIXUP_FUNC,
/* The GPIO must be pulled to initialize the AMP */
.v.func = alc_fixup_gpio4,
.chained = true,
- .chain_id = ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC
+ .chain_id = ALC294_FIXUP_SPK2_TO_DAC1
},
+
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7080,6 +7096,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE),
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, 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),
@@ -7167,7 +7185,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_GPIO),
+ 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, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
@@ -7643,11 +7661,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x1a, 0x90a70130},
{0x1b, 0x90170110},
{0x21, 0x03211020}),
- SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
- {0x12, 0xb7a60130},
- {0x13, 0xb8a61140},
- {0x16, 0x90170110},
- {0x21, 0x04211020}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x14, 0x90170110},
@@ -7841,6 +7854,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = {
SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
{0x19, 0x40000000},
{0x1a, 0x40000000}),
+ SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
+ {0x19, 0x40000000},
+ {0x1a, 0x40000000}),
{}
};
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index c80a16ee6e76..242542e23d28 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -647,6 +647,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
unsigned long flags;
unsigned char mclk_change;
unsigned int i, old_rate;
+ bool call_set_rate = false;
if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
return -EINVAL;
@@ -670,7 +671,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
* setting clock rate for internal clock mode */
old_rate = ice->get_rate(ice);
if (force || (old_rate != rate))
- ice->set_rate(ice, rate);
+ call_set_rate = true;
else if (rate == ice->cur_rate) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
return 0;
@@ -678,12 +679,14 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
}
ice->cur_rate = rate;
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+
+ if (call_set_rate)
+ ice->set_rate(ice, rate);
/* setting master clock */
mclk_change = ice->set_mclk(ice, rate);
- spin_unlock_irqrestore(&ice->reg_lock, flags);
-
if (mclk_change && ice->gpio.i2s_mclk_changed)
ice->gpio.i2s_mclk_changed(ice);
if (ice->gpio.set_pro_rate)
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index f4ee6798154a..7a5621e5e233 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/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index f8b5b960e597..4eaa2b5b20a5 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -292,7 +292,7 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
- uinfo->count = FIELD_SIZEOF(struct hdmi_codec_priv, eld);
+ uinfo->count = sizeof_field(struct hdmi_codec_priv, eld);
return 0;
}
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f6bf4cfbea23..e46b6ada13b1 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 57965cd678b4..a197114b0dad 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 3af36ec928e9..088b77931727 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 b1713fffa3eb..ae6f6121bc1b 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 7d7ea15d73e0..5ffbaddd6e49 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 3e5c69fbc33a..d9d59f45833f 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 10b82bf043d1..55e9f8800b3e 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 fbecbb74350b..68bcec5241f7 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 dd2b5ad08659..243f683bc02a 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 5d08ae066738..fb9ba8819706 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 61f230324164..6615ef64c7f5 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 062653ab03a3..1c84ff1a5bf9 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 76b7ee637e86..01e7bc03d92f 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 81d2af000a5c..b28613149b0c 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 2abf80b3eb52..92ef6a796fd5 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 9a9a381a908d..432d12bd4937 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 d82ab981e840..e20b806ec80f 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);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 2991b9986f66..395403a2d33f 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -145,6 +145,7 @@ struct snd_usb_substream {
struct snd_usb_endpoint *sync_endpoint;
unsigned long flags;
bool need_setup_ep; /* (re)configure EP at prepare? */
+ bool need_setup_fmt; /* (re)configure fmt after resume? */
unsigned int speed; /* USB_SPEED_XXX */
u64 formats; /* format bitmasks (all or'ed) */
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 9c8930bb00c8..a11c8150af58 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -506,15 +506,15 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
if (WARN_ON(!iface))
return -EINVAL;
alts = usb_altnum_to_altsetting(iface, fmt->altsetting);
- altsd = get_iface_desc(alts);
- if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
+ if (WARN_ON(!alts))
return -EINVAL;
+ altsd = get_iface_desc(alts);
- if (fmt == subs->cur_audiofmt)
+ if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt)
return 0;
/* close the old interface */
- if (subs->interface >= 0 && subs->interface != fmt->iface) {
+ if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) {
if (!subs->stream->chip->keep_iface) {
err = usb_set_interface(subs->dev, subs->interface, 0);
if (err < 0) {
@@ -528,6 +528,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
subs->altset_idx = 0;
}
+ if (subs->need_setup_fmt)
+ subs->need_setup_fmt = false;
+
/* set interface */
if (iface->cur_altsetting != alts) {
err = snd_usb_select_mode_quirk(subs, fmt);
@@ -1728,6 +1731,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
subs->data_endpoint->retire_data_urb = retire_playback_urb;
subs->running = 0;
return 0;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+ stop_endpoints(subs, true);
+ subs->need_setup_fmt = true;
+ return 0;
+ }
+ break;
}
return -EINVAL;
@@ -1760,6 +1770,13 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
subs->data_endpoint->retire_data_urb = retire_capture_urb;
subs->running = 1;
return 0;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+ stop_endpoints(subs, true);
+ subs->need_setup_fmt = true;
+ return 0;
+ }
+ break;
}
return -EINVAL;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 70c338f3ae24..d187aa6d50db 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3466,7 +3466,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
.vendor_name = "Dell",
.product_name = "WD19 Dock",
.profile_name = "Dell-WD15-Dock",
- .ifnum = QUIRK_NO_INTERFACE
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_FMT_AFTER_RESUME
}
},
/* MOTU Microbook II */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 349e1e52996d..a81c2066499f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -508,6 +508,16 @@ static int create_standard_mixer_quirk(struct snd_usb_audio *chip,
return snd_usb_create_mixer(chip, quirk->ifnum, 0);
}
+
+static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ struct usb_driver *driver,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ chip->setup_fmt_after_resume_quirk = 1;
+ return 1; /* Continue with creating streams and mixer */
+}
+
/*
* audio-interface quirks
*
@@ -546,6 +556,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
+ [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
};
if (quirk->type < QUIRK_TYPE_COUNT) {
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index ff3cbf653de8..6fe3ab582ec6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -33,7 +33,7 @@ struct snd_usb_audio {
wait_queue_head_t shutdown_wait;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
-
+ unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
int num_interfaces;
int num_suspended_intf;
int sample_rate_read_error;
@@ -98,6 +98,7 @@ enum quirk_type {
QUIRK_AUDIO_EDIROL_UAXX,
QUIRK_AUDIO_ALIGN_TRANSFER,
QUIRK_AUDIO_STANDARD_MIXER,
+ QUIRK_SETUP_FMT_AFTER_RESUME,
QUIRK_TYPE_COUNT
};