Through the examinations and experiments with lots of Roland and BOSS
USB-audio devices, we found out that the recently introduced
full-duplex operations with the implicit feedback mode work fine for
quite a few devices, while the others need only the capture-side quirk
to enforce the full-duplex mode. The recent commit d86f43b17e
("ALSA: usb-audio: Add support for many Roland devices' implicit
feedback quirks") tried to add such quirk entries manually in the
lists, but this turned out to be too many and error-prone, hence it
was reverted again.
This patch is another attempt to cover those missing Roland/BOSS
devices but in a more generic way. It matches the devices with the
vendor ID 0x0582, and checks whether they are with both ASYNC sync
types or ASYNC is only for capture device. In the former case, it's
the device with the implicit feedback mode, and applies accordingly.
In both cases, the capture stream requires always the full-duplex
mode, and we apply the known capture quirk for that, too.
Basically the already existing BOSS device quirk entries become
redundant after this generic matching, so those are removed. Although
the capture_implicit_fb_quirks[] table became empty and superfluous, I
keep it for now, so that people can put a special device easily at any
time later again.
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519
Tested-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/20210422120413.457-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit d86f43b17e ("ALSA: usb-audio: Add support for
many Roland devices' feedback quirks").
It turned out that many quirk entries there don't contain the proper
EP values and/or the quirk types, which lead to the broken
operations.
As we're going to cover all Roland/BOSS devices in a more generic way
rather the explicit lists, let's revert the previous additions at
first.
Fixes: d86f43b17e ("ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks")
Link: https://lore.kernel.org/r/20210422120413.457-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A reorganization of the driver source led to two of them causing
a compile time warning in some configurations:
tegra/tegra20_spdif.c:36:12: error: 'tegra20_spdif_runtime_resume' defined but not used [-Werror=unused-function]
36 | static int tegra20_spdif_runtime_resume(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra20_spdif.c:27:12: error: 'tegra20_spdif_runtime_suspend' defined but not used [-Werror=unused-function]
27 | static int tegra20_spdif_runtime_suspend(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra30_ahub.c:64:12: error: 'tegra30_ahub_runtime_resume' defined but not used [-Werror=unused-function]
64 | static int tegra30_ahub_runtime_resume(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~
tegra/tegra30_ahub.c:43:12: error: 'tegra30_ahub_runtime_suspend' defined but not used [-Werror=unused-function]
43 | static int tegra30_ahub_runtime_suspend(struct device *dev)
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~
Mark these functions as __maybe_unused to avoid this kind of warning.
Fixes: b5571449e6 ("ASoC: tegra30: ahub: Remove handing of disabled runtime PM")
Fixes: c53b396f0d ("ASoC: tegra20: spdif: Remove handing of disabled runtime PM")
Fixes: 80ec4a4cb3 ("ASoC: tegra20: i2s: Remove handing of disabled runtime PM")
Fixes: b5f6f781fc ("ASoC: tegra30: i2s: Remove handing of disabled runtime PM")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20210422133418.1757893-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Configuring number of channels per LRCLK frame by using e.g.
snd_soc_dai_set_tdm_slot before configuring DAI format was being
overwritten by the latter due to a regmap_write which would write over
the whole register.
Signed-off-by: Niklas Carlsson <niklasc@axis.com>
Link: https://lore.kernel.org/r/20210422130226.15201-1-Niklas.Carlsson@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Parse dai/tdm/clk are common for both CPU/Codec node.
This patch creates simple_parse_node() for it and share the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87czuoi41f.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Parse mclk_fs/dai/tdm/clk are common for both CPU/Codec node.
This patch creates graph_parse_node() for it and share the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fszki426.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Random noise could be heard when playing audio to the HDMI output.
This is due to the IEC conversion is invoked in the external loop.
As a result, this additional loop takes up a lot of the processing
cycle.
hdmi_reformat_iec958() process the conversion using an internal loop,
it is safe to move it out from the external loop to avoid unnecessary
processing cycle been spent. Furthermore, ALSA IEC958 plugin works in
32bit format only.
Signed-off-by: Sia Jee Heng <jee.heng.sia@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210421005546.7534-1-jee.heng.sia@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi Mark
These patches cleanups audio-graph.
This is part of prepare for new audio-graph-card2.
Kuninori Morimoto (6):
ASoC: audio-graph: move audio_graph_card_probe() to simple-card-utils.c
ASoC: audio-graph: move audio_graph_remove() to simple-card-utils.c
ASoC: audio-graph: check ports if exists
ASoC: audio-graph: remove "audio-graph-card," preix support
ASoC: audio-graph: remove unused "node" from graph_parse_mclk_fs()
ASoC: audio-graph: remove Platform support
include/sound/graph_card.h | 4 --
include/sound/simple_card_utils.h | 3 ++
sound/soc/generic/audio-graph-card.c | 52 ++++--------------------
sound/soc/generic/simple-card-utils.c | 25 ++++++++++++
sound/soc/generic/simple-card.c | 7 ----
sound/soc/tegra/tegra_audio_graph_card.c | 4 +-
6 files changed, 38 insertions(+), 57 deletions(-)
--
2.25.1
Thank you for your help !!
Best regards
---
Kuninori Morimoto
Enable Daisy Chain if in TDM mode and the number of played
channels is bigger than the maximum supported number of channels.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1618915453-29445-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Platform was one of mandatory component on ASoC before,
and audio-graph-card was assuming that CPU and Platform were
same driver.
But it is no longer mandatory on ASoC.
Current ASoC will just ignore if Platform and CPU were same
or doplicated component.
Of course ASoC is supporting Platform, but current
audio-graph-card doesn't support detecting it from DT.
This means current audio-graph-card operation for Platform so far
is 100% useless. This patch removes it.
We can respawn it when we need it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sg3n3ubg.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No upstream code is using "audio-graph-card," preix,
and Yaml base Document doesn't indicate it.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v98j3ubp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
"endpoint" and "port" are always exists, but there is no guarantee
for "ports". This patch checks "ports" if exists, otherwise,
it might set un-expected settings.
This patch also do align to 100 char in 1 line.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wnsz3ubu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card2 can reuse audio_graph_remove() / asoc_simple_remove().
This patch moves it to simple-card-utils.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2df3uby.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card2 can reuse audio_graph_card_probe().
This patch moves it to simple-card-utils.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zgxv3uc4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL)
the error return path returns -ENOMEM via the exit label "error". Other
uses of the same error exit label set the err variable to -ENOMEM but this
is not being used. I believe the original intent was for the error exit
path to return the value in err rather than the hard coded -ENOMEM, so
return this rather than the hard coded -ENOMEM.
Addresses-Coverity: ("Unused value")
Fixes: 738d9edcfd ("ALSA: usb-audio: Add sanity checks for invalid EPs")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210420134719.381409-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct link_info can grow fairly large and may cause the stack frame
size to be exceeded when allocated on the stack. Some architectures
such as 32-bit ARM, RISC-V or PowerPC have small stack frames where
this causes a compiler warning, so allocate these structures on the
heap instead of the stack.
Fixes: 343e55e718 ("ASoC: simple-card-utils: Increase maximum number of links to 128")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Reported-by: kernel test robot <lkp@intel.com>
Signed-off-by: Thierry Reding <treding@nvidia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20210419164117.1422242-1-thierry.reding@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On HP EliteBook 845 G8, the audio LEDs can be enabled by
ALC285_FIXUP_HP_MUTE_LED. So use it accordingly.
In addition to that, the mic captures lots of noises, so also limits the
mic boost. The quality of capture audio becomes crystal clear after
limiting the mic boost.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210420115530.1349353-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last two patches in this series fix a longstanding issue that prevented
the ALC3263 codec from using a headset mic. This codec can be found on Dell
systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343.
In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use
legacy HD Audio just to avoid this issue:
https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnUtp89iPn2d2Q@mail.gmail.com/
This may allow that ACPI quirk to be removed. Either way, the other systems
mentioned above do not support this quirk and already use the ASoC driver,
so this fix is necessary for headset mic support on those systems.
Note: there is likely other handling for this codec that only exists in the
HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824
("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that
does not seem to be resolved in the ASoC driver, to give an example.
Other patches in this series are not specific to the ALC3263. These patches
set the correct combo jack configuration when headphones are inserted, and
fix a misaligned value set in the DMIC2 Configuration Default register.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961
Signed-off-by: David Ward <david.ward@gatech.edu>
David Ward (5):
ASoC: rt286: Fix upper byte in DMIC2 configuration
ASoC: rt286: Configure combo jack for headphones
ASoC: rt298: Configure combo jack for headphones
ASoC: rt286: Make RT286_SET_GPIO_* readable and writable
ASoC: rt286: Generalize support for ALC3263 codec
sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++-------------
sound/soc/codecs/rt298.c | 9 +++++++--
2 files changed, 28 insertions(+), 15 deletions(-)
--
2.31.1
The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350
and Venue 11 Pro 7140. They require the same handling for the combo jack to
work with a headset: GPIO pin 6 must be set.
The HDA driver always sets this pin on the ALC3263, which it distinguishes
by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028
(Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to
determine if Dell is the system vendor.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961
Signed-off-by: David Ward <david.ward@gatech.edu>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
During jack detection, the combo jack is configured for a CTIA headset, and
then for an OMTP headset, while sensing the mic connection. If a mic is not
found in either case, the combo jack should be re-configured for headphones
only. This is consistent with the HDA driver behavior.
Signed-off-by: David Ward <david.ward@gatech.edu>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210418134658.4333-4-david.ward@gatech.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
During jack detection, the combo jack is configured for a CTIA headset, and
then for an OMTP headset, while sensing the mic connection. If a mic is not
found in either case, the combo jack should be re-configured for headphones
only. This is consistent with the HDA driver behavior.
Signed-off-by: David Ward <david.ward@gatech.edu>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210418134658.4333-3-david.ward@gatech.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
The ALSA control readback functionality only works for non-volatile
controls, i.e. control values that does not change on their own without
driver interaction.
This doesn't work for readbacks since the DSP firmware updates the
control value. Disable the cache mechanism in the driver if the control
name matches the prefix used for readbacks to ensure that the control
value is valid.
Signed-off-by: Niklas Carlsson <niklasc@axis.com>
Link: https://lore.kernel.org/r/20210419144901.9441-1-Niklas.Carlsson@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The last two patches in this series fix a longstanding issue that prevented
the ALC3263 codec from using a headset mic. This codec can be found on Dell
systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343.
In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use
legacy HD Audio just to avoid this issue:
https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnUtp89iPn2d2Q@mail.gmail.com/
This may allow that ACPI quirk to be removed. Either way, the other systems
mentioned above do not support this quirk and already use the ASoC driver,
so this fix is necessary for headset mic support on those systems.
Note: there is likely other handling for this codec that only exists in the
HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824
("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that
does not seem to be resolved in the ASoC driver, to give an example.
Other patches in this series are not specific to the ALC3263. These patches
set the correct combo jack configuration when headphones are inserted, and
fix a misaligned value set in the DMIC2 Configuration Default register.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961
Signed-off-by: David Ward <david.ward@gatech.edu>
David Ward (5):
ASoC: rt286: Fix upper byte in DMIC2 configuration
ASoC: rt286: Configure combo jack for headphones
ASoC: rt298: Configure combo jack for headphones
ASoC: rt286: Make RT286_SET_GPIO_* readable and writable
ASoC: rt286: Generalize support for ALC3263 codec
sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++-------------
sound/soc/codecs/rt298.c | 9 +++++++--
2 files changed, 28 insertions(+), 15 deletions(-)
--
2.31.1
base-commit: a38fd87484
This HDA verb sets the upper byte of the Configuration Default register, so
only an 8-bit value should be used. For the rt298, the same fix was applied
in commit f8f2dc4a71 ("ASoC: rt298: fix wrong setting of gpio2_en").
Signed-off-by: David Ward <david.ward@gatech.edu>
Link: https://lore.kernel.org/r/20210418134658.4333-2-david.ward@gatech.edu
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pioneer devices are supposed to be working with the implicit feedback
mode, but so far the attempt to apply the implicit feedback caused
issues, hence we explicitly skipped the implicit feedback mode for
them. Recently, Geraldo discovered that the device actually works if
you skip the generic matching of the sync EPs for the capture stream.
That is, we should apply the implicit feedback setup for the playback
like other similar devices, while we need to return 1 from
audioformat_capture_quirk() so that no further matching will be done.
And, later on, Olivia reported later that the fiddling with the
capture quirk alone doesn't suffice for the test with speaker-test
program. This seems to be a similar case like the recently fixed BOSS
devices. Indeed, the problem could be addressed by setting
playback_first flag, which indicates that the playback URBs have to be
sent out at first even in the implicit feedback mode.
This patch implements the application of the implicit feedback to
Pioneer devices as described in the above. The former
skip_pioneer_sync_ep() was dropped, and instead we provide
is_pioneer_implicit_fb() to check the Pioneer devices that need the
implicit feedback. In the audioformat_implicit_fb_quirk(), simply
apply the implicit fb for playback and set chip->playback_first flag
if matching, and in audioformat_capture_quirk()(), it returns 1 for
skipping the generic EP sync handling.
Reported-by: Geraldo <geraldogabriel@gmail.com>
Tested-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de
Link: https://lore.kernel.org/r/20210419153918.450-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cppcheck warning:
sound/soc/codecs/rt5682.c:2404:42: style: Boolean result is used in
bitwise operation. Clarify expression with
parentheses. [clarifyCondition]
(pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST));
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210416191144.27006-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
sound/soc/codecs/lpass-rx-macro.c:1626:9: warning: Identical condition
and return expression 'ret', return value is always 0
[identicalConditionAfterEarlyExit]
return ret;
^
sound/soc/codecs/lpass-rx-macro.c:1623:6: note: If condition 'ret' is
true, the function will return/exit
if (ret)
^
sound/soc/codecs/lpass-rx-macro.c:1626:9: note: Returning identical
expression 'ret'
return ret;
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210416191144.27006-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
sound/soc/soc-core.c:2784:6: style: Condition '!num_routes' is always
false [knownConditionTrueFalse]
if (!num_routes) {
^
sound/soc/soc-core.c:2777:17: note: Assuming that condition
'num_routes<0' is not redundant
if (num_routes < 0 || num_routes & 1) {
^
sound/soc/soc-core.c:2783:2: note: Compound assignment '/=', assigned
value is 0
num_routes /= 2;
^
sound/soc/soc-core.c:2784:6: note: Condition '!num_routes' is always
false
if (!num_routes) {
^
The documentation for of_property_count_string reads
"
* Returns the number of strings on
* success, -EINVAL if the property does not exist, -ENODATA if property
* does not have a value, and -EILSEQ if the string is not null-terminated
* within the length of the property data.
"
Since the case for num_routes == 0 is not possible, let's remove this
test.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210416191144.27006-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
sound/soc/soc-core.c:2237:13: warning: %x in format string (no. 1)
requires 'unsigned int *' but the argument type is 'signed
int *'. [invalidScanfArgType_int]
} else if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
^
sound/soc/soc-core.c:2237:13: warning: %x in format string (no. 2)
requires 'unsigned int *' but the argument type is 'signed
int *'. [invalidScanfArgType_int]
} else if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210416191144.27006-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
cppcheck warning:
value that is never used. [unreadVariable]
acpi_status status = AE_OK;
^
sound/soc/soc-acpi.c:37:21: style: Variable 'status' is assigned a
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210416191144.27006-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add case statement to set sample-rate for the DJM-750 Pioneer
mixer. This was included as part of another patch but I think it has
been archived on Patchwork and hasn't been merged.
Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210418165901.25776-1-livvy@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GA503 has almost exactly the same default setup as the GA401
model with the same issues. The GA401 quirks solve all the issues
so we will use the full quirk chain.
Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210419030411.28304-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes USB audio capture and playback possible and pristine on my Roland
INTEGRA-7, Boutique D-05, and R-26, along with many more I've encountered
people having had issues with over the last decade or so.
Signed-off-by: Lucas Endres <jaffa225man@gmail.com>
Link: https://lore.kernel.org/r/CAOsVg8rA61B=005_VyUwpw3piVwA7Bo5fs1GYEB054efyzGjLw@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add main capture switch and main capture volume control.
Main capture control has its own channel value respectivelly.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/dfd43a8db04e4d52a889d6f5c1262173@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On Tegra186 and later, the number of links can go up to 72, so bump the
maximum number of links to the next power of two (128).
Fixes: f2138aed23 ("ASoC: simple-card-utils: enable flexible CPU/Codec/Platform")
Signed-off-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20210416071147.2149109-2-thierry.reding@gmail.com
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI counting code doesn't propagate errors when the number of
maximum links is exceeded, which causes subsequent initialization code
to continue to run and that eventually leads to memory corruption with
the code trying to access memory that is out of bounds.
Fix this by propagating errors when the maximum number of links is
reached, which ensures that the driver fails to load and prevents the
memory corruption.
Fixes: f2138aed23 ("ASoC: simple-card-utils: enable flexible CPU/Codec/Platform")
Signed-off-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20210416071147.2149109-1-thierry.reding@gmail.com
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp
used for ELD notifications), display connection change done during suspend,
can be lost due to following sequence of events:
1. system in S3 suspend
2. DP/HDMI receiver connected
3. system resumed
4. HDA controller resumed, but card->deferred_resume_work not complete
5. acomp eld_notify callback
6. eld_notify ignored as power state is not CTL_POWER_D0
7. HDA resume deferred work completed, power state set to CTL_POWER_D0
This results in losing the notification, and the jack state reported to
user-space is not correct.
The check on step 6 was added in commit 8ae743e82f ("ALSA: hda - Skip
ELD notification during system suspend"). It would seem with the deferred
resume logic in ASoC core, this check is not safe.
Fix the issue by modifying the check to use "dev.power.power_state.event"
instead of ALSA specific card power state variable.
BugLink: https://github.com/thesofproject/linux/issues/2825
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_pcm_params_symmetry() checks CPU / Codec symmetry.
Unfortunately there was bug on it (= A) which didn't check Codec.
But is back by (B).
A: v5.7: commit c840f7698d ("ASoC: soc-pcm: Merge for_each_rtd_cpu/codec_dais()")
B: v5.12: commit 3a90672111 ("ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()")
In total,
old - v5.6 (= Generation-1):
symmetric_rate : DAI_Link / CPU / Codec
symmetric_channels : DAI_Link / CPU / Codec
symmetric_sample_bits : DAI_Link / CPU / Codec
v5.7 - v5.11 (= Generation-2): (= because of bug by (A))
symmetric_rate : DAI_Link / CPU
symmetric_channels : DAI_Link / CPU / Codec
symmetric_sample_bits : DAI_Link / CPU / Codec
v5.12 - (= Generation-3): (= back by (B))
symmetric_rate : DAI_Link / CPU / Codec
symmetric_channels : DAI_Link / CPU / Codec
symmetric_sample_bits : DAI_Link / CPU / Codec
OTOH, we can use DPCM which is configured by FE / BE.
Both FE / BE uses dummy-DAI.
FE: CPU <-> dummy-DAI
BE: dummy-DAI <-> Codec
One note is that we can use .be_hw_params_fixup in DPCM case.
This means BE settings might be fixuped/updated by FE.
This feature is used for example on MIXer case.
It can be happen not only for rate, but for channels/sample_bits too.
Because of these reasons, below issue happen on
Generation-1 / Generation-3, if...
1) Sound Card used DPCM
2) It exchanges rate to 48kHz by using .be_hw_params_fixup()
3) Codec had symmetric_rate = 1
I didn't confirm, but maybe same things happen
if it exchanged channels/sample_bits at Generation-1/2/3 too.
# aplay 44100.wav
# aplay 44100.wav
=> [kernel] be.ak4613-hifi: ASoC: unmatched rate symmetry: snd-soc-dummy-dai:44100 - soc_pcm_params_symmetry:48000
[kernel] be.ak4613-hifi: ASoC: hw_params BE failed -22
[kernel] fe.rsnd-dai.0: ASoC: hw_params BE failed -22
aplay: set_params:1407: Unable to install hw params:
ACCESS: RW_INTERLEAVED
FORMAT: S16_LE
SUBFORMAT: STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: 44100
PERIOD_TIME: (23219 23220)
PERIOD_SIZE: 1024
PERIOD_BYTES: 4096
PERIODS: 4
BUFFER_TIME: (92879 92880)
BUFFER_SIZE: 4096
BUFFER_BYTES: 16384
TICK_TIME: 0
soc_pcm_params_symmetry() checks by below
if (symmetry)
for_each_rtd_cpu_dais(rtd, i, cpu_dai)
if (cpu_dai->xxx && cpu_dai->xxx != d.xxx) {
dev_err(rtd->dev, "...");
return -EINVAL;
}
Because of above reason 3) (= Codec had symmetric_rate = 1)
BE can't ignore "if (symmetric)".
At 1st aplay, soc_pcm_params_symmetry() ignores it,
because dummy-DAI->rate is 0.
After this check, each DAI sets/keep settings.
In above sample case, BE gets 48000 and FE gets 44100,
and it happen BE -> FE order.
Because DPCM is sharing *same* dummy-DAI,
dummy-DAI sets as 48000 by BE, and is overwrote by 44100 by FE.
This settings never be cleaned (= a) after 1st aplay,
because dummy-DAI is used from FE/BE, never be last user (b).
static int soc_pcm_hw_clean(...)
{
...
for_each_rtd_dais(rtd, i, dai) {
...
(b) if (snd_soc_dai_active(dai) == 1)
(a) soc_pcm_set_dai_params(dai, NULL);
...
}
...
}
At 2nd aplay, BE gets 48000 but dummy-DAI is keeping 44100,
soc_pcm_params_symmetry() checks will fail.
To solve this issue, this patch ignores dummy-DAI
at soc_pcm_params_symmetry()
Link: https://lore.kernel.org/r/87a6q0z4xt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2djxa2n.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It indicates unmatched symmetry value, but not indicates on which DAI.
This patch indicates it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rbbyono.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
__soc_pcm_params_symmetry() macro is using "name" as parameter
which will be exchanged to rate/channles/sample_bit, like below
dai->name => dai->rate
dai->name => dai->channels
dai->name => dai->sample_bit
But, dai itself has "name". This means
1) It is very confusable naming
2) It can't use dai->name
This patch use "xxx" instead of "name"
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8735vryoob.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A DAI link has 3 components:
* CPU
* platform
* codec(s)
A component is specified via:
* name
* of_node
* dai_name
In order to avoid confusion when building a sound card we disallow
matching by both name and of_node (1).
soc_check_tplg_fes allows overriding certain BE links by overriding
BE link name. This doesn't work well if BE link was specified via DT,
because we end up with a link with both name and of_node specified
which is conflicting with (1).
In order to fix this we need to:
* override of_node if component was specified via DT
* override name, otherwise.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20210414101212.65573-1-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks for jack detection, rt711 DAI and DMIC
Reviewed-by: Bard Liao <bard.liao@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210415175013.192862-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add one configuration with no RT711.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Link: https://lore.kernel.org/r/20210415175013.192862-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add configurations ported over from TGL.
The topology names need to include link information given all the
hardware permutations.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Link: https://lore.kernel.org/r/20210415175013.192862-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix two bugs with the Intel HDA Realtek ALC233 sound codec
present in Intel NUC NUC8i7BEH and probably a few other similar
NUC models.
These codecs advertise a 4-level microphone input boost amplifier on
pin 0x19, but the highest two boost settings do not work correctly,
and produce only low analog noise that does not seem to contain any
discernible signal. There is an existing fixup for this exact problem
but for a different PCI subsystem ID, so we re-use that logic.
Changing the boost level also triggers a DC spike in the input signal
that bleeds off over about a second and overwhelms any input during
that time. Thankfully, the existing fixup has the side effect of
making the boost control show up in userspace as a mute/unmute switch,
and this keeps (e.g.) PulseAudio from fiddling with it during normal
input volume adjustments.
Finally, the NUC hardware has built-in inverted stereo mics. This
patch also enables the usual fixup for this so the two channels cancel
noise instead of the actual signal.
[ Re-ordered the quirk entry point by tiwai ]
Signed-off-by: Phil Calvin <phil@philcalvin.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/80dc5663-7734-e7e5-25ef-15b5df24511a@philcalvin.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a report about the possible race in the user control element
counts (card->user_ctl_count), and it was confirmed that the race
wasn't serious in the old code up to 5.12. There, the value
modification itself was exclusive and protected via a write semaphore,
hence it's at most concurrent reads and evaluations before the
increment. Since it's only about the soft-limit to avoid the
exhausting memory usage, one-off isn't a big problem at all.
Meanwhile, the relevant code has been largely modified recently, and
now card->user_ctl_count was replaced with card->user_ctl_alloc_size,
and a few more places were added to access this field. And, in this
new code, it turned out to be more serious: the modifications are
scattered in various places, and a few of them are without protection.
It implies that it may lead to an inconsistent value by racy
accesses.
For addressing it, this patch extends the range covered by the
card->controls_rwsem write lock at snd_ctl_elem_add() so that the all
code paths that modify and refer to card->user_ctl_alloc_size are
protected by the rwsem properly.
The patch adds also comments in a couple of functions to indicate that
they are under the rwsem lock.
Fixes: 66c6d1ef86 ("ALSA: control: Add memory consumption limit to user controls")
Link: https://lore.kernel.org/r/FEEBF384-44BE-42CF-8FB3-93470933F64F@purdue.edu
Link: https://lore.kernel.org/r/20210415131856.13113-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently, the sof_pcm_dai_link_fixup() function was
updated to match SSP config with the PCM hw_params
and set the current_config for the DAI widget.
But the sof_restore_pipelines() function still chooses the
default config for the DAI widget upon resuming. Fix this
to use the last used config when setting up the DAI widget
during resume.
Fixes: c943a586f6 ("ASoC: SOF: match SSP config with pcm hw params")
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210415162107.130963-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.
The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.
This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.
Tested for all use cases of the driver.
Based on similar fix in kbl_rt5663_rt5514_max98927.c
from Harsha Priya <harshapriya.n@intel.com> and
Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Cc: <stable@vger.kernel.org> # 5.4+
Signed-off-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210415124347.475432-1-lma@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hi Mark
These patches adjusts to multi CPU/Codec on simple-card / audio-graph.
This is part of prepare for new audio-graph-card2.
Kuninori Morimoto (5):
ASoC: simple-card: remove unused variable from simple_parse_of()
ASoC: simple-card: use asoc_link_to_xxx() macro
ASoC: simple-card: use simple_props_to_xxx() macro
ASoC: audio-graph: use asoc_link_to_xxx() macro
ASoC: audio-graph: use simple_props_to_xxx() macro
sound/soc/generic/audio-graph-card.c | 43 +++++++++++++-----------
sound/soc/generic/simple-card.c | 50 +++++++++++++---------------
2 files changed, 47 insertions(+), 46 deletions(-)
--
2.25.1
We shouldn't use dai_props->cpus/codecs directly,
because these are array now to supporting multi CPU/Codec/Platform.
This patch uses simple_props_to_xxx() macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87k0p5zs97.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't use dai_link->cpus/codecs/platforms directly,
because these are array now to supporting multi CPU/Codec/Platform.
This patch uses asoc_link_to_xxx() macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87lf9lzs9c.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't use dai_props->cpus/codecs directly,
because these are array now to supporting multi CPU/Codec/Platform.
This patch uses simple_props_to_xxx() macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mtu1zs9i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't use dai_link->cpus/codecs/platforms directly,
because these are array now to supporting multi CPU/Codec/Platform.
This patch uses asoc_link_to_xxx() macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o8ehzs9n.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit d9ffff696c ("ASoC: simple-card: Use snd_soc_of_parse_aux_devs()")
switched to use snd_soc_of_parse_aux_devs() on simple_parse_of().
Thus noone is using *top anymore. Let's cleanup unused variable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pmyxzs9w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are few instances of KUNIT tests that are not properly defined.
This commit focuses on correcting these issues to match the standard
defined in the Documentation.
Issues Fixed:
- tests should end in KUNIT_TEST, some fixes have been applied to
correct issues were KUNIT_TESTS is used or KUNIT is not mentioned.
- Tests should default to KUNIT_ALL_TESTS
- Tests configs tristate should have if !KUNIT_ALL_TESTS
No functional changes other than CONFIG name changes
Changes since v2:
- Split patch 1 by subcomponents
- fix issues where config was *KUNIT_TEST_TEST
- properly threaded/chained messages
Nico Pache (6):
kunit: ASoC: topology: adhear to KUNIT formatting standard
kunit: software node: adhear to KUNIT formatting standard
kunit: ext4: adhear to KUNIT formatting standard
kunit: lib: adhear to KUNIT formatting standard
kunit: mptcp: adhear to KUNIT formatting standard
m68k: update configs to match the proper KUNIT syntax
arch/m68k/configs/amiga_defconfig | 6 +++---
arch/m68k/configs/apollo_defconfig | 6 +++---
arch/m68k/configs/atari_defconfig | 6 +++---
arch/m68k/configs/bvme6000_defconfig | 6 +++---
arch/m68k/configs/hp300_defconfig | 6 +++---
arch/m68k/configs/mac_defconfig | 6 +++---
arch/m68k/configs/multi_defconfig | 6 +++---
arch/m68k/configs/mvme147_defconfig | 6 +++---
arch/m68k/configs/mvme16x_defconfig | 6 +++---
arch/m68k/configs/q40_defconfig | 6 +++---
arch/m68k/configs/sun3_defconfig | 6 +++---
arch/m68k/configs/sun3x_defconfig | 6 +++---
drivers/base/test/Kconfig | 2 +-
drivers/base/test/Makefile | 2 +-
fs/ext4/.kunitconfig | 2 +-
fs/ext4/Kconfig | 2 +-
fs/ext4/Makefile | 2 +-
lib/Kconfig.debug | 21 +++++++++++++--------
lib/Makefile | 6 +++---
net/mptcp/Kconfig | 2 +-
net/mptcp/Makefile | 2 +-
net/mptcp/crypto.c | 2 +-
net/mptcp/token.c | 2 +-
sound/soc/Kconfig | 2 +-
sound/soc/Makefile | 4 ++--
25 files changed, 64 insertions(+), 59 deletions(-)
--
2.30.2
Current ssi.c has duplicated code to control BUSIF
over/under run interrupt.
This patch adds new rsnd_ssi_busif_err_irq_enable/disable()
and share the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Yongbo Zhang <giraffesnn123@gmail.com>
Cc: Chen Li <licheng0822@thundersoft.com>
Link: https://lore.kernel.org/r/871rbl1jsb.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ssi.c clears BUSIF error status at __rsnd_ssi_interrupt(),
but its code is verbose.
This patch off-load it to rsnd_ssi_busif_err_status_clear().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8735w11jso.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 66c705d07d ("SoC: rsnd: add interrupt support for SSI BUSIF
buffer") adds __rsnd_ssi_interrupt() checks for BUSIF status,
but is using "break" at for loop.
This means it is not checking all status. Let's check all BUSIF status.
Fixes: commit 66c705d07d ("SoC: rsnd: add interrupt support for SSI BUSIF buffer")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874kgh1jsw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd needs to call .prepare (P) for clock settings,
.trigger for playback start (S) and stop (E).
It should be called as below from SSI point of view.
P -> S -> E -> P -> S -> E -> ...
But, if you used MIXer, below case might happen
(2)
1: P -> S ---> E -> ...
2: P ----> S -> ...
(1) (3)
P(1) setups clock, but E(2) resets it. and starts playback (3).
In such case, it will reports "SSI parent/child should use same rate".
rsnd_ssi_master_clk_start() which is the main function at (P)
was called from rsnd_ssi_init() (= S) before,
but was moved by below patch to rsnd_soc_dai_prepare() (= P) to avoid
using clk_get_rate() which shouldn't be used under atomic context.
commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set
under non-atomic")
Because of above patch, rsnd_ssi_master_clk_start() is now called at (P)
which is for non atomic context. But (P) is assuming that spin lock is
*not* used.
One issue now is rsnd_ssi_master_clk_start() is checking ssi->xxx
which should be protected by spin lock.
After above patch, adg.c had below patch for other reasons.
commit 06e8f5c842 ("ASoC: rsnd: don't call clk_get_rate()
under atomic context")
clk_get_rate() is used at probe() timing by this patch.
In other words, rsnd_ssi_master_clk_start() is no longer using
clk_get_rate() any more.
This means we can call it from rsnd_ssi_init() (= S) again which is
protected by spin lock.
This patch re-move it to under spin lock, and solves
1. checking ssi->xxx without spin lock issue.
2. clk setting / device start / device stop race condition.
Reported-by: Linh Phung T. Y. <linh.phung.jy@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875z0x1jt5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During the recent rewrite of the implicit feedback support, we've
tested to apply the implicit fb on BOSS devices, but it failed, as the
capture stream didn't start without the playback. As the end result,
it got another type of quirk for tying both streams but starts
playback always (commit 6234fdc1ce "ALSA: usb-audio: Quirk for BOSS
GT-001").
Meanwhile, Mike Oliphant has tested the real implicit feedback mode
for the playback again with the latest code, and found out that it
actually works if the initial feedback sync is skipped; that is, on
those BOSS devices, the playback stream has to be started at first
without waiting for the capture URB completions. Otherwise it gets
stuck. In the rest operations after the capture stream processed, we
can take them as the implicit feedback source.
This patch is an attempt to improve the support for BOSS devices with
the implicit feedback mode in the way described above. It adds a new
flag to snd_usb_audio, playback_first, indicating that the playback
stream starts without sync with the initial capture completion. This
flag is set in the quirk table with the new IMPLICIT_FB_BOTH type.
Reported-and-tested-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210414083255.9527-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a bad idea to allocate big structures on the stack.
Mark the variables as static and add a note for the locking.
Fixes: 22d8de62f1 ("ALSA: control - add generic LED trigger module as the new control layer")
Cc: Nathan Chancellor <nathan@kernel.org>
Cc: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20210414105858.1937710-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi Mark
This patch-set is for v2 of Multi-CPU/Codec support,
and some cleanups.
v1 had patch-conflict on simple-card / audio-graph with below.
v2 was solved it.
fa74c223b6
("ASoC: simple-card: fix possible uninitialized single_cpu local variable")
I want to add new audio-graph-card2 driver which can support
not only DPCM, but also Multi-CPU/Codec, and Codec2Codec.
And it is also supporting audio-graph-card2 base custom driver.
But before supporting such driver, we need to cleanup existing
simple-card / audio-graph, because these and new driver are
sharing code.
Link: https://lore.kernel.org/r/87wntmod33.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/20210408141639.GA39604@sirena.org.uk
Kuninori Morimoto (12):
ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing
ASoC: simple-card-utils: remove li->dais/li->conf
ASoC: simple-card-utils: use for_each_prop_xxx()
ASoC: simple-card-utils: remove asoc_simple_parse_xxx()
ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference()
ASoC: simple-card-utils: indicate dai_fmt if exist
ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug
ASoC: simple-card-utils: add simple_props_to_xxx() macro
ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform()
ASoC: simple-card-utils: tidyup debug info for clock
ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line
ASoC: simple-card-utils: tidyup asoc_simple_parse_convert()
include/sound/simple_card_utils.h | 107 ++++++++++------
sound/soc/generic/audio-graph-card.c | 64 ++++------
sound/soc/generic/simple-card-utils.c | 174 ++++++++++++++++----------
sound/soc/generic/simple-card.c | 70 ++++-------
4 files changed, 226 insertions(+), 189 deletions(-)
--
2.25.1
This patch adds missing MODULE_DEVICE_TABLE definition which generates
correct modalias for automatic loading of this driver when it is built
as an external module.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Chen Lifu <chenlifu@huawei.com>
Reviewed-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Link: https://lore.kernel.org/r/20210409015953.259688-1-chenlifu@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current asoc_simple_canonicalize_cpu/platform() is assuming single CPU,
single Platform, but we want to support Multi support.
This patch is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87im4swf8y.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC is now supporting multi DAI, but, current
simple-card / audio-graph are assuming fixed single DAI.
This patch cares multi DAI at asoc_simple_clean_reference()
for of_node_put().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o8ekwf9p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC is now supporting multi DAI, but, current
simple-card / audio-graph are assuming fixed single DAI.
Now, asoc_simple_parse_xxx() macro is assuming single DAI.
To support multi-CPU/Codec, this patch unpack asoc_simple_parse_xxx()
macro, and uses "&dai_link->cpus[i]" instead of "dai_link->cpus".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pmz0wf9u.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC is now supporting multi DAI, but, current
simple-card / audio-graph are assuming fixed single DAI.
This patch uses for_each_prop_xxx() to support multi DAI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r1jgwf9y.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
li->dais is same as number of CPU + Codec,
li->conf is same as number of Codec when dummy-Codec.
li->dais/li->conf are no longer needed.
This patch removes these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sg3wwfa3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We couldn't setup dai_props cpu_dai/codec_dai at the initial timing,
because "counting DAIs loop" and "detecting DAIs loop" were different.
But we can do it now, because these are using same loops.
This patch setups dai_props cpu_dai/codec_dai at the initial timing.
It can removes triky code from simple-card / audio-graph.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tuocwfa8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Previous fifo depth patch was only tested on axg, not g12 or sm1.
Of course, while adding hw_params dai callback for the axg, I forgot to do
the same for g12 and sm1, leaving the depth unset and breaking playback on
these SoCs.
Add hw_params callback to the g12 dai_ops to fix the problem.
Fixes: 6f68accaa8 ("ASoC: meson: axg-frddr: set fifo depth according to the period")
Reported-by: Christian Hewitt <christianshewitt@gmail.com>
Tested-by: Christian Hewitt <christianshewitt@gmail.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20210412132256.89920-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the current code, we have some assumption that the audio clock
selector has been set up implicitly and don't want to touch it unless
it's really needed for the fallback autoclock setup. This works for
most devices but some seem having a problem. Partially this was
covered for the devices with a single connector at the initialization
phase (commit 086b957cc1 "ALSA: usb-audio: Skip the clock selector
inquiry for single connections"), but also there are cases where the
wrong clock set up is kept silently. The latter seems to be the cause
of the noises on Behringer devices.
In this patch, we explicitly set up the audio clock selector whenever
the appropriate node is found.
Reported-by: Geraldo Nascimento <geraldogabriel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327
Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxxdw@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210413084152.32325-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One missed property for TigerLake and need for separate descriptors
between ADL-S and the other flavors.
Libin Yang (1):
ASoC: SOF: Intel: add missing use_acpi_target_states for TGL platforms
Sathya Prakash M R (1):
ASoC: SOF: Intel: Update ADL P to use its own descriptor
sound/soc/sof/intel/pci-tgl.c | 20 +++++++++++++++++++-
sound/soc/sof/intel/tgl.c | 2 +-
2 files changed, 20 insertions(+), 2 deletions(-)
--
2.25.1
Takashi Iwai