Takashi Iwai [Tue, 5 Jun 2018 14:51:55 +0000 (16:51 +0200)]
Merge tag 'asoc-v4.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.18
This is a very big update, mainly due to a huge set of new drivers some
of which are individually very large. We also have a lot of fixes for
the topology stuff, several of the users have stepped up and fixed some
the serious issues there, and continued progress on the transition away
from CODEC specific drivers to generic component drivers.
- Many fixes for the topology code, including fixes for the half done
v4 ABI compatibility from Guenter Roeck and other ABI fixes from
Kirill Marinushkin.
- Lots of cleanup for Intel platforms based on Realtek CODECs from Hans
de Goode.
- More followups on removing legacy CODEC things and transitioning to
components from Morimoto-san.
- Conversion of OMAP DMA to the new, more standard SDMA-PCM driver.
- A series of fixes and updates to the rather elderly Cirrus Logic SoC
drivers from Alexander Sverdlin.
- Qualcomm DSP support from Srinivas Kandagatla.
- New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306 and
RT5668 and TI TSCS454
Guenter Roeck [Thu, 24 May 2018 19:49:21 +0000 (12:49 -0700)]
ASoC: topology: Improve backwards compatibility with v4 topology files
Commit dc31e741db49 ("ASoC: topology: ABI - Add the types for BE
DAI") introduced sound topology files version 5. Initially, this
change made the topology code incompatible with v4 topology files.
Backwards compatibility with v4 configuration files was
subsequently added with commit 288b8da7e992 ("ASoC: topology:
Support topology file of ABI v4").
Unfortunately, backwards compatibility was never fully implemented.
First, the manifest size in (Skylake) v4 configuration files is set
to 0, which causes manifest_new_ver() to bail out with error messages
similar to the following.
snd_soc_skl 0000:00:1f.3: ASoC: invalid manifest size
snd_soc_skl 0000:00:1f.3: tplg component load failed-22
snd_soc_skl 0000:00:1f.3: Failed to init topology!
snd_soc_skl 0000:00:1f.3: ASoC: failed to probe component -22
skl_n88l25_m98357a skl_n88l25_m98357a: ASoC: failed to instantiate card -22
skl_n88l25_m98357a: probe of skl_n88l25_m98357a failed with error -22
After this problem is fixed, the following error message is seen instead.
snd_soc_skl 0000:00:1f.3: ASoC: old version of manifest
snd_soc_skl 0000:00:1f.3: Invalid descriptor token 1093938482
snd_soc_skl 0000:00:1f.3: ASoC: failed to load widget media0_in cpr 0
snd_soc_skl 0000:00:1f.3: tPlg component load failed-22
This message is seen because backwards compatibility for loading widgets
was never implemented.
The lack of audio support when running the upstream kernel on recent
Chromebooks has been reported in various forums, and can be traced back
to this problem. Attempts to fix the problem, usually by providing v5
configuration files, were only partially successful.
Let's implement backward compatibility properly to solve the problem
for good.
Signed-off-by: Guenter Roeck <groeck@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Ben Hutchings [Fri, 1 Jun 2018 15:32:30 +0000 (16:32 +0100)]
ALSA: pci/hda: Remove unused, broken, header file
sound/pci/hda/local.h seems to be an earlier version of
sound/hda/local.h; it was added at the same time but doesn't seem to
have ever been used (within the git history). Most of its macros
depend on a hdac_read_parm() function which is not defined anywhere.
Signed-off-by: Ben Hutchings <ben.hutchings@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mac Chiang [Thu, 31 May 2018 17:18:32 +0000 (01:18 +0800)]
ASoC: Intel: kbl: Move codec sysclk config to codec_init function
On APL, commit fd0f237572ad
("ASoC: Intel: bxt: Move codec sysclk config to codec_init function")
fixed an issue related to jack detection.
The MCLK for DA7219 does not change in this platform, but is
currently being configured everytime as part of the platform_clock
event handler for DAPM. The upshot of this is that we have
unnecessary calls to this function, and it also means that if
a stream hasn't yet been started, DA7219 driver does not have the
correct MCLK rates programmed and so the HP detection feature does
not operate as expected.
The same fix is needed on KBL.
This patch rectifies this issue by moving the sysclk call to
codec_init function so it's only called once at initialisation.
Signed-off-by: Mac Chiang <mac.chiang@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Daniel Mack [Wed, 30 May 2018 19:45:56 +0000 (21:45 +0200)]
ASoC: simple-card: set cpu dai clk in hw_params
The simple-card driver currently accepts a clock node in the cpu dai
sub-node and only uses it as an alternative to the
'system-clock-frequency' property to get the current frequency.
This patch adds another use of the passed clock node. If mclk-fs is
specified, the clocks in cpu and codec dai sub-nodes will be set to
the calculated rate (stream rate * mclk_fs) in hw_params.
This allows platforms to pass tuneable clocks as phandle that will
automatically be set to the right rates.
Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Bo Chen [Thu, 31 May 2018 22:35:18 +0000 (15:35 -0700)]
ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
When 'kzalloc()' fails in 'snd_hda_attach_pcm_stream()', a new pcm instance is
created without setting its operators via 'snd_pcm_set_ops()'. Following
operations on the new pcm instance can trigger kernel null pointer dereferences
and cause kernel oops.
This bug was found with my work on building a gray-box fault-injection tool for
linux-kernel-module binaries. A kernel null pointer dereference was confirmed
from line 'substream->ops->open()' in function 'snd_pcm_open_substream()' in
file 'sound/core/pcm_native.c'.
This patch fixes the bug by calling 'snd_device_free()' in the error handling
path of 'kzalloc()', which removes the new pcm instance from the snd card before
returns with an error code.
Signed-off-by: Bo Chen <chenbo@pdx.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Xie Yisheng [Thu, 31 May 2018 11:11:23 +0000 (19:11 +0800)]
ASoC: dapm: use match_string() helper
match_string() returns the index of an array for a matching string,
which can be used instead of open coded variant.
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Fri, 18 May 2018 19:35:06 +0000 (21:35 +0200)]
ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks
Many X86 devices using a BYT SoC + RT5651 codec are cheap devices with
generic DMI strings, causing snd_soc_set_dmi_name() to fail to set a
long_name, making it impossible for userspace to have a correct UCM
profile which knowns which input is connected to the internal mic,
which input is connected to the hsmic (for correct jack-based switching)
and which inputs are unused.
Our quirks already specify which inputs the internal and headset mic
are connected to.
This commit sets a long_name based on the quirks so that userspace can
have UCM profiles doing the right thing based on the long_name.
Note that if we ever encounter the need for a special UCM profile for
some device we can add a quirk to set a specific long_name for the
device,
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes: b3c702f56bf5 ("ASoC: mt6797: combine DAI to register component") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Wed, 30 May 2018 21:53:45 +0000 (23:53 +0200)]
ASoC: codecs: PCM1789: include gpio/consumer.h
When CONFIG_GPIOLIB is disabled, this codec fails to build
because gpio/consumer.h is not included implicitly.
sound/soc/codecs/pcm1789.c: In function 'pcm1789_common_init':
sound/soc/codecs/pcm1789.c:247:19: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_gpio_request_one'? [-Werror=implicit-function-declaration]
pcm1789->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_HIGH);
^~~~~~~~~~~~~~~~~~~~~~~
Fixes: 4ae340d1be36 ("ASoC: codecs: Add support for PCM1789") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Dan Carpenter [Thu, 31 May 2018 06:25:07 +0000 (09:25 +0300)]
ALSA: xen-front: fix a loop timeout
We want the loop to exit when "to" is set to zero, but in the current
code it's set to -1. Also I tweaked the indenting so it doesn't look
like we're passing "--to" to xenbus_read_unsigned().
Jon Hunter [Wed, 30 May 2018 15:15:19 +0000 (16:15 +0100)]
ASoC: core: Fix return code shown on error for hw_params
When the call to hw_params for a component fails, the error code is held
by the variable '__ret' but the error message displays the value held by
the variable 'ret'. Fix the return code shown when hw_params fails for
a component.
Fixes: b8135864d4d3 ("ASoC: snd_soc_component_driver has snd_pcm_ops") Signed-off-by: Jon Hunter <jonathanh@nvidia.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Philipp Zabel [Wed, 23 May 2018 09:43:00 +0000 (11:43 +0200)]
ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
In asynchronous mode, a RxFS and RxClk connection needs to be made between
two ports. Add a define for the bit to be set in the *SEL fields.
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
[m.felsch@pengutronix.de: fixed comment to include i.MX21 and 35] Signed-off-by: Marco Felsch <m.felsch@pengutronix.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Pierre-Louis Bossart [Tue, 29 May 2018 23:30:02 +0000 (18:30 -0500)]
ASoC: fix 0-day warnings with snd_soc_new_compress()
All conditionally-defined routines in include/sound/soc.h expose a
static inline fallback to avoid 0-day warnings and compilation issues,
except snd_soc_new_compress().
Fixes: 5db6aab6f36f ('ASoC: topology: Add support for compressed PCMs') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hui Wang [Wed, 30 May 2018 04:33:07 +0000 (12:33 +0800)]
ALSA: hda/realtek - Enable mic-mute hotkey for several Lenovo AIOs
We have several Lenovo AIOs like M810z, M820z and M920z, they have
the same design for mic-mute hotkey and led and they use the same
codec with the same pin configuration, so use the pin conf table to
apply fix to all of them.
Fixes: 29693efcea0f ("ALSA: hda - Fix micmute hotkey problem for a lenovo AIO machine") Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tom Briden [Tue, 29 May 2018 16:34:20 +0000 (17:34 +0100)]
ALSA: hda/realtek - Fixup for HP x360 laptops with B&O speakers
Added a new helper file for these fixups due to requiring a huge number
of coefs being set to get the top speakers to work, as well as
setting pin 0x17 for the top speakers and the correct input source
of 0x17 for volume control
[ Note: this is a revised work based on Tom's fixup patch with the
replacement of the full COEF tables provided by Realtek.
Also, the fixup function has a proper HDA_FIXUP_ACT_* handling now.
The credit for the new COEF table goes to Kailang -- tiwai ]
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189331 Cc: Kailang Yang <kailang@realtek.com> Signed-off-by: Tom Briden <tom@decompile.me.uk> Tested-by: Tom Briden <tom@decompile.me.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Srinivas Kandagatla [Tue, 29 May 2018 10:18:30 +0000 (11:18 +0100)]
ASoC: qdsp6: q6afe-dai: use q6afe_dai_prepare() for MI2S
Use common q6afe_dai_prepare() for MI2S dais, this will remove
some code duplication. Also make the if statement to switch to
make the code look neater.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Chieh Chuang [Mon, 28 May 2018 02:18:19 +0000 (10:18 +0800)]
ASoC: dpcm: symmetry constraint on FE substream
We should set BE symmetric constraint on FE substream.
in case one BE is used by two FE1/FE2,
the first BE runtime will use FE1's substream->runtime.
hence the FE1's will be constrained by BE symmetry property.
Though, second FE2 call dpcm_apply_symmetry,
the be_substream->runtime == FE1's substream->runtime.
The FE2's substream->runtime will not be constrained
by BE's symmetry property.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Chieh Chuang [Mon, 28 May 2018 02:18:18 +0000 (10:18 +0800)]
ASoC: dpcm: fix BE dai not hw_free and shutdown
In case, one BE is used by two FE1/FE2
FE1--->BE-->
|
FE2----]
when FE1/FE2 call dpcm_be_dai_hw_free() together
the BE users will be 2 (> 1), hence cannot be hw_free
the be state will leave at, ex. SND_SOC_DPCM_STATE_STOP
later FE1/FE2 call dpcm_be_dai_shutdown(),
will be skip due to wrong state.
leaving the BE not being hw_free and shutdown.
The BE dai will be hw_free later when calling
dpcm_be_dai_shutdown() if still in invalid state.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Agrawal, Akshu [Mon, 28 May 2018 03:48:22 +0000 (11:48 +0800)]
ASoC: AMD: make channel 1 dma as circular
channel 1: SYSMEM<->ACP
channel 2: ACP<->I2S
Instead of waiting on period interrupt of ch 2 and then starting
dma on ch1, we make ch1 dma as circular.
This removes dependency of period granularity on hw pointer.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Tested-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Mon, 28 May 2018 20:26:49 +0000 (22:26 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for the ARCHOS 80 Cesium 8" windows tablet
Add a quirk for the ARCHOS 80 Cesium 8" windows tablet, this device mostly
works with the default settings, except that it has only one speaker.
So add a quirk with the default settings + the mono-speaker flag.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Tom Briden [Sat, 25 Mar 2017 10:12:01 +0000 (10:12 +0000)]
ALSA: hda/realtek - Fixup mute led on HP Spectre x360
This patch adds the mute LED control for HP Spectre x360 Kabylake
model. The mute LED is controlled via VREF bits on NID 0x1b, so we
need a new fixup function.
Note that this doesn't fix the other issues like the missing speaker
output on the machine. They will be addressed by later patches.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189331 Signed-off-by: Tom Briden <tom@decompile.me.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sun, 27 May 2018 11:01:17 +0000 (13:01 +0200)]
ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers
Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as
this serves merely as an intermediate buffer that is copied to each
URB transfer buffer. This works well in general on x86, but on some
archs this may result in cache coherency issues when mmap is used.
OTOH, it works also on such arch unless mmap is used.
This patch is a step for mitigating the inconvenience; a new module
option "use_vmalloc" is provided so that user can choose to allocate
the DMA coherent buffer instead of the existing vmalloc buffer.
The drawback is that it'd be the standard dma_alloc_coherent() calls
and the system would require contiguous pages on non-x86 archs.
Note that it's a global option and not dynamically switchable since
the buffer is pre-allocated at the probe time. In theory, it's
possible to be switchable, but it'd be trickier and racier.
As default use_vmalloc option is set to true, so that the old behavior
is kept. For allowing the coherent mmap on ARM or MIPS, pass
use_vmalloc=0 option explicitly.
Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Arnd Bergmann [Mon, 28 May 2018 15:59:57 +0000 (17:59 +0200)]
ALSA: xen: ensure nul-terminated device name
gcc-8 warns that pcm_instance->name is not necessarily terminated correctly
if the input is more than 80 characters long or lacks a termination byte
itself:
In function 'strncpy',
inlined from 'cfg_device' at sound/xen/xen_snd_front_cfg.c:399:3,
inlined from 'xen_snd_front_cfg_card' at sound/xen/xen_snd_front_cfg.c:509:9:
include/linux/string.h:254:9: error: '__builtin_strncpy' specified bound 80 equals destination size [-Werror=stringop-truncation]
return __builtin_strncpy(p, q, size);
Using strlcpy() instead of strncpy() makes this a bit safer.
Takashi Iwai [Sun, 27 May 2018 13:07:01 +0000 (15:07 +0200)]
ALSA: usb-audio: Avoid lowlevel device object
Simplify the device management by replacing the lowlevel device object
allocation with the card->private_data. Nowadays there is almost no
advantage by the lowlevel device, and with card->private_data, the
code becomes cleaner.
The stream direction in open and close callbacks can be retrieved from
substream->direction, hence we don't have to stick with the unique PCM
ops hard-coded for each direction. Rewrite the common open/close
callback functions.
Colin Ian King [Sun, 27 May 2018 21:32:19 +0000 (22:32 +0100)]
ALSA: xen-front: fix unsigned error check on return from to_sndif_format
The negative error return from the call to to_sndif_format is being
assigned to an unsigned 8 bit integer and hence the check for a negative
value is always going to be false. Fix this by using ret as the error
return and hence the negative error can be detected and assign
the u8 sndif_format to ret if there is no error.
Detected by CoverityScan, CID#1469385 ("Unsigned compared against 0")
Signed-off-by: Colin Ian King <colin.king@canonical.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamoccchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Colin Ian King [Sun, 27 May 2018 21:23:12 +0000 (22:23 +0100)]
ALSA: xen-front: remove redundant error check on ret
The error for a -ve value in ret is redundant as all previous
assignments to ret have an associated -ve check and hence it
is impossible for ret to be less that zero at the point of the
check. Remove this redundant error check.
Detected by CoveritScan, CID#1469407 ("Logically Dead code")
Signed-off-by: Colin Ian King <colin.king@canonical.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 27 May 2018 01:13:30 +0000 (10:13 +0900)]
ALSA: dice: unuse second stream for MIDI conformant data channel for TC Electronic models
At present, all of models produced by TC Electronic except for Konnekt Live
are supported with hard-coded their stream formats. Studio Konnekt 48 is
sore model to support dual streams for both directions. The second stream
has no MIDI conformant data channel in its data block. But current
implementation transfers the second stream with MIDI conformant data
channel.
Takashi Sakamoto [Sun, 27 May 2018 01:13:29 +0000 (10:13 +0900)]
ALSA: dice: fix stream format parameters for TC Electronic Studio Konnekt 48
TC Electronic Studio Konnekt 48 is an application of combination of
WaveFront Dice II STD and TC Applied Technologies (TCAT) TCD2210 (Dice
Mini). The latter is on a board with BNC and optical interfaces, thus
used for signal processing for word clock, S/PDIF and ADAT. This model
doesn't support TCAT extended application protocol. For such devices,
ALSA dice driver needs to have hard-coded parameters for stream formats.
This commit fixes stream format parameters for this model. Unfortunately, at
sampling transmission frequencies over 48.0kHz, I confirmed that current
ALSA dice driver doesn't drive the device appropriately to generate sounds
(silence). I guess that this comes from timestamping quirk of Dice-based
devices, which I reported.
[alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6
http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html
$ cd linux-firewire-utils/src
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 04044a26 bus_info_length 4, crc_length 4, crc 18982
404 31333934 bus_name "1394"
408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255,
max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400)
40c 00016604 company_id 000166 |
410 08a65810 device_id 0408a65810 | EUI-64 0001660408a65810
root directory
-----------------------------------------------------------------
414 00062ab9 directory_length 6, crc 10937
418 03000166 vendor
41c 8100000a --> descriptor leaf at 444
420 17000022 model
424 8100000f --> descriptor leaf at 460
428 0c0087c0 node capabilities per IEEE 1394
42c d1000001 --> unit directory at 430
unit directory at 430
-----------------------------------------------------------------
430 0004d5c5 directory_length 4, crc 54725
434 12000166 specifier id
438 13000001 version
43c 17000022 model
440 8100000f --> descriptor leaf at 47c
By simply allow build testing without DMA_OMAP, we can shut up that warning.
Fixes: dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Lukas Wunner [Thu, 24 May 2018 17:01:07 +0000 (19:01 +0200)]
ALSA: hda - Fix runtime PM
Before commit 3b5b899ca67d ("ALSA: hda: Make use of core codec functions
to sync power state"), hda_set_power_state() returned the response to
the Get Power State verb, a 32-bit unsigned integer whose expected value
is 0x233 after transitioning a codec to D3, and 0x0 after transitioning
it to D0.
The response value is significant because hda_codec_runtime_suspend()
does not clear the codec's bit in the codec_powered bitmask unless the
AC_PWRST_CLK_STOP_OK bit (0x200) is set in the response value. That in
turn prevents the HDA controller from runtime suspending because
azx_runtime_idle() checks that the codec_powered bitmask is zero.
Since commit 3b5b899ca67d, hda_set_power_state() only returns 0x0 or
0x1, thereby breaking runtime PM for any HDA controller. That's because
an inline function introduced by the commit returns a bool instead of a
32-bit unsigned int. The change was likely erroneous and resulted from
copying and pasting snd_hda_check_power_state(), which is immediately
preceding the newly introduced inline function. Fix it.
Link: https://bugs.freedesktop.org/show_bug.cgi?id=106597 Fixes: 3b5b899ca67d ("ALSA: hda: Make use of core codec functions to sync power state") Cc: Alex Deucher <alexander.deucher@amd.com> Cc: Abhijeet Kumar <abhijeet.kumar@intel.com> Reported-and-tested-by: Gunnar Krüger <taijian@posteo.de> Signed-off-by: Lukas Wunner <lukas@wunner.de> Acked-by: Alex Deucher <alexander.deucher@amd.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 24 May 2018 09:20:06 +0000 (11:20 +0200)]
ALSA: echoaudio: Drop superfluous macro
Drop pci_device() macro that just leads to chip->pci->dev, and pass it
directly to request_firmware(). It was introduced for allowing the
external alsa-driver kernel module builds. Since it was discontinued
years ago, we should clean it up now.
Takashi Iwai [Thu, 24 May 2018 09:15:45 +0000 (11:15 +0200)]
ALSA: usb-audio: Drop superfluous ifndef
Drop the superfluous #ifndef checks that had been put just for
allowing building the alsa-driver kernel modules externally.
Since the external build was discontinued years ago, let's clean up
the old kludges.
Takashi Iwai [Thu, 24 May 2018 09:14:13 +0000 (11:14 +0200)]
ALSA: memalloc: Drop superfluous ifndef
Drop the superfluous #ifndef check in memalloc.h that had been put
just for allowing building the alsa-driver kernel modules externally.
Since the external build was discontinued years ago, let's clean up
the old kludges.
Hans de Goede [Wed, 23 May 2018 13:27:03 +0000 (15:27 +0200)]
ALSA: hda: Add Clevo W35xSS_370SS to the power_save blacklist
Power-saving is causing a plop and silences the first 2 seconds
(give or take) of audio, silencing notifications sounds on Medion /
Clevo W35xSS_370SS laptops.
Add the Clevo W35xSS_370SS to the power_save blacklist.
Hans de Goede [Mon, 21 May 2018 12:42:51 +0000 (14:42 +0200)]
ASoC: Intel: cht_bsw_nau8824: Fix jack_type to include SND_JACK_MICROPHONE
The nau8824 codec can detect whether a headset or plain headphones is
inserted (as well as button presses on the headset) as such the jack_type
passed to snd_soc_card_jack_new() should include SND_JACK_MICROPHONE.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Lin Huang [Tue, 22 May 2018 08:53:42 +0000 (16:53 +0800)]
ASoC: rockchip: cdn-dp sound output use spdif
some monitors care about the parity bit in the sub-frame of I2S,
but the cdn-dp always set this bit to "1", so these monitors
do not have sound output if use i2s, use spdif can fix this issue.
Signed-off-by: Chris Zhong <zyw@rock-chips.com> Signed-off-by: Lin Huang <hl@rock-chips.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Daniel Mack [Mon, 21 May 2018 21:50:16 +0000 (23:50 +0200)]
ASoC: pxa-ssp: allow more flexible setup order
The pxa-ssp driver currently assumes that .set_fmt() is called before
.set_clkdiv(), .set_pll() etc.
Commit a8bd0ee558714 ("ASoC: raumfeld: Use static DAI format setup") broke
support for Raumfeld hardware (and possible other PXA based ones) because
it effectively changed the order of these calls. Also, as the call to
.set_fmt() is now done at probe time, the port clock is not yet enabled.
To fix this, strip all hardware register access code from the .set_fmt()
callback and memorize the desired value, so we can use it from the
.hw_params() callback. Also make the .set_fmt() callback less destructive
by reading all registers that it writes to in the beginning and only
masking out the bits that it possibly fiddles with.
Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Srinivas Kandagatla [Fri, 18 May 2018 12:56:01 +0000 (13:56 +0100)]
ASoC: qdsp6: q6asm: Add q6asm driver
This patch adds basic support to Q6 ASM (Audio Stream Manager) module on
Q6DSP. ASM supports up to 8 concurrent streams. each stream can be setup
as playback/capture. ASM provides top control functions like
Pause/flush/resume for playback and record. ASM can Create/destroy encoder,
decoder and also provides POPP dynamic services.
This patch adds support to basic features to allow hdmi playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Srinivas Kandagatla [Fri, 18 May 2018 12:56:00 +0000 (13:56 +0100)]
ASoC: qdsp6: q6adm: Add q6adm driver
This patch adds support to Q6ADM (Audio Device Manager) module in
q6dsp. ADM performs routing between audio streams and AFE ports.
It does Rate matching for streams going to devices driven by
different clocks, it handles volume ramping, Mixing with channel
and bit-width. ADM creates and destroys dynamic COPP services
for device-related audio processing as needed.
This patch adds basic support to ADM.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Mukunda, Vijendar [Tue, 8 May 2018 04:47:53 +0000 (10:17 +0530)]
ASoC: amd: dma driver changes for bt i2s instance
With in ACP, There are three I2S controllers can be
configured/enabled ( I2S SP, I2S MICSP, I2S BT).
Default enabled I2S controller instance is I2S SP.
This patch provides required changes to support I2S BT
controller Instance.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Akshu Agrawal [Tue, 8 May 2018 04:47:50 +0000 (10:17 +0530)]
ASoC: AMD: Move clk enable from hw_params/free to startup/shutdown
hw_param can be called multiple times and thus we can have
more clk enable. The clk may not get diabled due to refcounting.
startup/shutdown ensures single clk enable/disable call.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Mukunda, Vijendar [Tue, 8 May 2018 04:47:49 +0000 (10:17 +0530)]
ASoC: amd: memory release for rtd structure
rtd structure freed early may result in kernel panic in dma close
call back. moved releasing memory for rtd structure to the end of
dma close callback.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Mukunda, Vijendar [Tue, 8 May 2018 04:47:48 +0000 (10:17 +0530)]
ASoC: amd: sram bank update changes
Added sram bank variable to audio_substream_data structure.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Mukunda, Vijendar [Tue, 8 May 2018 04:47:47 +0000 (10:17 +0530)]
ASoC: amd: pte offset related dma driver changes
Added pte offset variable in audio_substream_data structure.
Added Stoney related PTE offset macros in acp header file.
Modified hw_params callback to assign the pte offset value
based on asic_type.
PTE Offset macros used to calculate no of PTE entries
need to be programmed when memory allocated for audio buffer.
Depending upon allocated audio buffer size, PTE offset values
will change.
Compared to CZ, Stoney has SRAM memory limitation i.e 48k
It is required to define separate PTE Offset macros for
Stoney.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
projects / users / willy / xarray.git / log