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The ALC5651 does not like multi-write accesses, avoid them. This fixes:
rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121
Errors on resume (and all registers after the registers in the error not
being synced).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
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Enable 5-band graphic equalizer which allows manipulation of
gain in range -11.75dB to 12dB for 5 specific sound bands:
115Hz, 330Hz, 990Hz, 3000Hz, 9900Hz.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add mux for choosing DAC source: ADC, I2S or DAP. DAP - digital
audio processing block. Moreover create choosing mux for DAP main
source (ADC or I2S) and DAP mixer source (ADC or I2S). Mixer block
allows summing DAP main and mixer sources togheter with defined
volumes.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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With P0 R29, Bit2 set the I2S clocks will be running when the DAC/ADC is
powered down, but still the codec need to be powered up by needing at least
one complete DAPM path for the stream.
If the AIF is not needed (analog loopback for example) the I2S clocks will
not run as they are not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Suggested-by: Stefan Müller-Klieser <s.mueller-klieser@phytec.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The audio interface is not really the DAC input. Use more generic name for
it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now we can replace Codec to Component. Let's do it.
Note:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now we can replace Codec to Component. Let's do it.
Note:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now we can replace Codec to Component. Let's do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Now we can replace Codec to Component. Let's do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When we have same register to tell capture and playback capability
of a device and we want separate cpu dais for playback and
capture. Then, DW_I2S_QUIRK_COMP_PARAM1 is used to enable one
capability per dai.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The driver is used for AMD board using DA7219 and MAX98357 codec.
TEST=Build, apply grunt_mixer_settings:
iotools mmio_write32 0xfed80e40 0x040c40c0
iotools mmio_write32 0xfed80e28 0x10000
iotools mmio_write8 0xfed816de 0xc0
aplay -D plughw:0,0 ./sample_48000.wav -vv
aplay -D plughw:0,0 ./sample_44100.wav -vv
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-sgtl5000
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base and sysclk_reg should be unsigned int, as that is what
regmap takes as a register address.
sysclk_mask is used to mask unsigned register values so
should be unsigned.
sysclk_shift is a shift value so is not allowed to be negative.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in
SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume
sequence:
Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Stereo
aplay: pcm_write:2051: write error: Input/output error
The problem is caused by the fact that the aforementioned commit
dropped the cache handling, so re-introduce the register map
resync to fix the problem.
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
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Fixes: 08660086eff9 ("ASoC: ak4458: Add support for AK4458 DAC driver")
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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AK5558 is a 32-bit, 768 kHZ sampling, differential input ADC
for digital audio systems.
Datasheet is available at:
https://www.akm.com/akm/en/file/datasheet/AK5558VN.pdf
Initial patch includes support for normal and TDM modes.
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp>
[initial coding for 3.18 kernel]
Signed-off-by: Mihai Serban <mihai.serban@nxp.com>
[cleanups and porting to 4.9 kernel]
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
[tdm support]
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
[pm support, cleanups and porting to latest kernel]
Signed-off-by: Mark Brown <broonie@kernel.org>
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The AK4458 is a 32-bit 8ch Premium DAC that corresponds
to a 768kHz PCM input and an 11.2MHz DSD input at maximum.
It supports I2S, DSD and TDM modes with 24 or 32 bit MSB
or 16, 24, 32 LSB formats. Its datasheet is available here:
https://www.akm.com/akm/en/file/datasheet/AK4458VN.pdf
Signed-off-by: Junichi Wakasugi <wakasugi.jb@om.asahi-kasei.co.jp>
Signed-off-by: Mihai Serban <mihai.serban@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila@nxp.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Adds Kabylake kbl_da7219_max98357a_i2s machine driver entry into
machine table
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Kabylake-y I2S machine driver which uses
DA7219 as headset on ssp1 and MAX98357A codec as speaker
on ssp0.
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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One of the recently introduced commits hadn't been build tested, fix
that.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is some duplicate code in soc-core.c, and the kernel provides
DEFINE_SHOW_ATTRIBUTE() helper macro to decrease it in seq_file.h.
Signed-off-by: Peng Donglin <dolinux.peng@gmail.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Don't populate the const array norm_addrs on the stack, instead make it
static. Makes the object code smaller by over 230 bytes. Also re-format
array data as the insertion of the static keywork made the first line
overly long.
Before:
text data bss dec hex filename
53780 34752 256 88788 15ad4 linux/sound/soc/codecs/tscs42xx.o
After:
text data bss dec hex filename
53461 34840 256 88557 159ed linux/sound/soc/codecs/tscs42xx.o
(gcc version 7.2.0 x86_64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The check of blk_id == AUD_CLK_IO is redundant as it also being performed
in the following switch statement with the same return of -ENOTSUPP. Fix
this by removing the redundant comparison.
Detected by CoverityScan, CID#1465227 ("Logically dead code")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In usecases like hot plug-unplug DP panel or modeset during a playback,
sometimes we observe no audio after codec resets. During no audio
condition, we have noticed that the power state of the pin or the
connector is D3. Optimizing the way we set the power mitigates the
issue. With this changes the verb is sent to set the power state and
waits until actual state reaches target state. Thus ensuring power
state is set.
Signed-off-by: Abhijeet Kumar <abhijeet.kumar@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-intel
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{ NULL } only clears the first member of the structure.
Even though the first member of the snd_soc_dapm_update struct is a
pointer,it is more robust to use the empty struct initializer that
clears all the struct members.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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{ 0 } only clears the first member of the structure.
The first member of the snd_soc_dapm_update struct is a pointer,
and writing 0 to a pointer results in a sparse warning.
Use the empty struct initializer that clears all the struct members
and fixes the sparse warning.
Cc: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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{ 0 } only clears the first member of the structure.
The first member of the snd_soc_dapm_update struct is a pointer,
and writing 0 to a pointer results in a sparse warning.
Use the empty struct initializer that clears all the struct members
and fixes the sparse warning.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-sgtl5000
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Current implementation mute codec in global way (DAC block).
That means when user routes sound not from I2S but from
AUX source (LINE_IN) it also will be muted by alsa core.
This should not happen.
Signed-off-by: Michal Oleszczyk <oleszczyk.m@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In the reset state of the codec we do not have complete playback or capture
routes.
The audio playback/capture will not work due to missing clock signals on
the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down.
To make sure that even if all output/input is disconnected the codec is
generating clocks, we need to have valid DAPM route in every case to power
up the must needed parts of the codec.
I have verified that switching DAC (during playback) or ADC (during
capture) will stop the I2S clocks, so the only solution is to connect the
'Off' routes as well to output/input.
The routes will be only added if the codec is clock master. In case the
role changes runtime, the applied routes are removed.
Tested on am43x-epos-evm with aic3111 codec in master mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch defines I2S1 - HDMI DAI link and implements related
hw_params callback. The AUD PLL frequency is configured through
the CLK_SCLK_I2S1 leaf clock, the exynos5433 clock tree
definitions are updated in a separate patch.
The device tree parsing part is changed is a way it supports older
DTBs with just a single CPU DAI specified, without the HDMI link.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The SAMSUNG_I2S_OPCLK is not currently used by any card driver thus we can
safely change semantics of 'dir' argument of the I2S set_sysclk() callback.
Now an enumeration is exported instead of directly using register bit field
values.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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It seems both PCLK_I2S1 and SCLK_I2S1 clocks need to be enabled before
I2S1 control registers can be accessed on exynos5433. If SCLK clock
is disabled an exception is triggered. To fix this parent clock of
the RCLK_SRC clock is assigned to pri_dai->op_clk so required gate
clock is handled by the runtime PM ops.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In order to support multiple instances of the I2S IP block the platform
device name is prepended to each clock registered by the driver.
The clock-output-names property is now not used, this should not cause
any issues as, for example, CDCLK clock is referenced through DT 'clocks'
property, not by name.
This change allows to have both I2S0 and I2S1 enabled simultaneously
on exynos5433 and working properly when #clock-cells property is specified
in respective DT nodes.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-samsung
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Current soc_probe_dai() is using deep nested condition.
Thus, it is difficult to read/understand.
This patch simplification it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The dma_mapping_error() returns true or false, but we want to return
-ENOMEM if there was an error.
Fixes: e98131222ff9 ("ASoC: uniphier: add support for UniPhier AIO compress audio")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Tested-by: Katsuhiro Suzuki <suzuki.katsuhiro@socionext.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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{ 0 } only clears the first member of the structure.
The first member of the snd_soc_dapm_update struct is a pointer,
and writing 0 to a pointer results in the following sparse warning:
sound/soc/codecs/tlv320aic3x.c:175:47: warning: Using plain integer as NULL pointer
Use the empty struct initializer that clears all the struct members
and fixes the sparse warning.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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{ 0 } only clears the first member of the structure.
The first member of the snd_soc_dapm_update struct is a pointer,
and writing 0 to a pointer results in the following sparse warning:
sound/soc/codecs/wm9712.c:229:47: warning: Using plain integer as NULL pointer
Use the empty struct initializer that clears all the struct members
and fixes the sparse warning.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time"
decision. Then, it collects each component driver's .use_pmdown_time
settings by using AND operation.
But, ak4613 has no reason to ignore/use pmdown time.
Thus, It should has use_pmdown_time=0, it means it doesn't have
decision right of "ignore pmdown time".
Otherwise, ak4642 paired driver's pmdown_time settings doesn't
have meaning. In other words, "ignore pmdown time" is determined
irrespective of the value of paired driver's settings
(= ignore will be 0 by ak4642).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_soc_runtime_ignore_pmdown_time() returns "ignore pmdown time"
decision. Then, it collects each component driver's .use_pmdown_time
settings by using AND operation.
But, ak4613 has no reason to ignore/use pmdown time.
Thus, It should has use_pmdown_time=0, it means it doesn't have
decision right of "ignore pmdown time".
Otherwise, ak4613 paired driver's pmdown_time settings doesn't
have meaning. In other words, "ignore pmdown time" is determined
irrespective of the value of paired driver's settings
(= ignore will be 0 by ak4613).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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These laptops have a combined jack to attach headsets, the U727 on
the left, the U757 on the right, but a headsets microphone doesn't
work. Using hdajacksensetest I found that pin 0x19 changed the
present state when plugging the headset, in addition to 0x21, but
didn't have the correct configuration (shown as "Not connected").
So this sets the configuration to the same values as the headphone
pin 0x21 except for the device type microphone, which makes it
work correctly. With the patch the configured pins for U727 are
Pin 0x12 (Internal Mic, Mobile-In): present = No
Pin 0x14 (Internal Speaker): present = No
Pin 0x19 (Black Mic, Left side): present = No
Pin 0x1d (Internal Aux): present = No
Pin 0x21 (Black Headphone, Left side): present = No
Signed-off-by: Jan-Marek Glogowski <glogow@fbihome.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now we can replace Codec to Component. Let's do it.
Because there are many drivers which are using arizona,
we need to update these all related drivers in same time.
Otherwise compile error/warning happen
Note:
cs47l24
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 1 -> .idle_bias_on = 0
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
wm5102
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 1 -> .idle_bias_on = 0
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
wm5110
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 1 -> .idle_bias_on = 0
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
wm8997
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 1 -> .idle_bias_on = 0
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
wm8998
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 1 -> .idle_bias_on = 0
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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ALSA sequencer core initializes the event pool on demand by invoking
snd_seq_pool_init() when the first write happens and the pool is
empty. Meanwhile user can reset the pool size manually via ioctl
concurrently, and this may lead to UAF or out-of-bound accesses since
the function tries to vmalloc / vfree the buffer.
A simple fix is to just wrap the snd_seq_pool_init() call with the
recently introduced client->ioctl_mutex; as the calls for
snd_seq_pool_init() from other side are always protected with this
mutex, we can avoid the race.
Reported-by: 范龙飞 <long7573@126.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Audigy 2 CA0102 chip (but most likely others from the emu10k1 family,
too) has a problem that from time to time it likes to do few DMA reads a
bit beyond its normal allocation and gets very confused if these reads get
blocked by a IOMMU.
For the first (reserved) page this happens multiple times at every
playback, for various synth pages it happens randomly, rarely for PCM
playback buffers and the page table memory itself.
All these reads seem to follow a similar pattern, observed read offsets
beyond the allocation end were 0x00, 0x40, 0x80 and 0xc0 (PCI cache line
multiples), so it looks like the device tries to accesses up to 256 extra
bytes.
As a workaround let's widen these DMA allocations by an extra page if we
detect that the device is behind a non-passthrough IOMMU (the DMA memory
should be relatively plenty on IOMMU systems).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit a5003fc04113 ("[ALSA] emu10k1 - simplify page allocation for synth")
switched from using the DMA allocator for synth DMA pages to manually
calling alloc_page().
However, this usage has an implicit assumption that the DMA address space
for the emu10k1-family chip is the same as the CPU physical address space
which is not true for a system with a IOMMU.
Since this made the synth part of the driver non-functional on such systems
let's effectively revert that commit (while keeping the
__synth_free_pages() simplification).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When we get a IOMMU page fault for a emu10k1 device it is very hard to
discover which of chip many DMA allocations triggered it (since on a IOMMU
system the DMA address space is often very different from the CPU one).
Let's add optional debug printouts providing this information.
These debug printouts are only enabled on an explicit request via the
kernel dynamic debug mechanism.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have been calling dma_set_mask() and then dma_set_coherent_mask() with
the same value, but there is a dma_set_mask_and_coherent() function that
does exactly that so let's use it instead.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The emu10k1-family chips need the first page (index 0) reserved in their
page tables for some reason (every emu10k1 driver I've checked does this
without much of an explanation).
Using the first page for normal samples results in a broken playback.
However, we already have a dummy page allocated - so called "silent page"
and, in fact, had always been setting it as the first page in the chip page
table because an initialization of every entry of the page table to point
to a silent page happens after and overwrites the reserved_page allocation.
So the only thing remaining to remove the reserved_page allocation is a
trivial change to the page allocation logic to ignore the first page entry
and start its allocations from the second entry (index 1).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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