Wang, Xiaoming [Thu, 12 Jun 2014 22:47:07 +0000 (18:47 -0400)]
ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform.
Cancel the optimization of compiler for struct snd_compr_avail
which size will be 0x1c in 32bit kernel while 0x20 in 64bit
kernel under the optimizer. That will make compaction between
32bit and 64bit. So add packed to fix the size of struct
snd_compr_avail to 0x1c for all platform.
Bios does not set up the pin config default correctly (everything
is set to zero). Reporter claims that 6stack-dig and 6stack-automute
solve the problem.
Adam Goode [Wed, 4 Jun 2014 15:20:55 +0000 (11:20 -0400)]
ALSA: seq: Continue broadcasting events to ports if one of them fails
Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.
How to reproduce problem:
*** termA
$ aseqdump -p 0:1
0:1 Port subscribed 0:1 -> 128:0
*** termB
$ aseqdump -p 0:1
*** termA
0:1 Client start client 129
0:1 Port start 129:0
0:1 Port subscribed 0:1 -> 129:0
*** termB
0:1 Port subscribed 0:1 -> 129:0
*** termA
^C
*** termB
0:1 Client exit client 128
<--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently mutex_destroy() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
ALSA: fireworks: Remove a constant over width to which it's applied
The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type.
But this member is 1 byte. Although the value is between 0x00-0xff, a constant
has 0x10000. This constant is meaningless.
Adam Goode [Wed, 4 Jun 2014 05:02:51 +0000 (01:02 -0400)]
ALSA: seq: correctly detect input buffer overflow
snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 3 Jun 2014 10:31:46 +0000 (12:31 +0200)]
ASoC: Fix wrong argument for card remove callbacks
The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type. The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.
Fixes: e1d4d3c854f2 ('ASoC: free jack GPIOs before the sound card is freed') Acked-by: Mark Brown <broonie@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 3 Jun 2014 09:51:14 +0000 (11:51 +0200)]
Merge tag 'asoc-v3.16-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.16
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
Stephen Warren [Fri, 30 May 2014 18:42:57 +0000 (12:42 -0600)]
ASoC: free jack GPIOs before the sound card is freed
This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.
That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========
Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Mark Brown [Mon, 2 Jun 2014 15:08:21 +0000 (16:08 +0100)]
ASoC: cache: Fix error code when not using ASoC level cache
It is not an error to have no cache so we shouldn't return an error code
and cause our callers to fail, just silently do nothing instead. Thanks
to Jarkko for identify the problematic commit.
Reported-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Reported-by: Fabio Estevam <festevam@gmail.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Takashi Iwai [Mon, 2 Jun 2014 13:16:07 +0000 (15:16 +0200)]
ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
The conversion to a fixup table for Replacer model with ALC260 in
commit 20f7d928 took the wrong widget NID for COEF setups. Namely,
NID 0x1a should have been used instead of NID 0x20, which is the
common node for all Realtek codecs but ALC260.
Fixes: 20f7d928fa6e ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser') Cc: <stable@vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ronan Marquet [Sun, 1 Jun 2014 16:38:53 +0000 (18:38 +0200)]
ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
Correcion of wrong fixup entries add in commit ca8f0424 to replace
static model quirk for PB V7900 laptop (will model).
[note: the removal of ALC260_FIXUP_HP_PIN_0F chain is also needed as a
part of the fix; otherwise the pin is set up wrongly as a headphone,
and user-space (PulseAudio) may be wrongly trying to detect the jack
state -- tiwai]
ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio
data means Valid Length Code (VBL). Ths value is:
- b00 for 24 bits sample (label is 0x40)
- b01 for 20 bits sample (label is 0x41)
- b10 for 16 bits sample (label is 0x42)
But current firewire-lib apply 24 bits label for both of 16/24 bits samples.
As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them
have a behaviour to ignore the label. They can generate correct sound even
if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only
for Raw Audio data channel, but also for IEC 60958 Conformant data channel.
Oder Chiou [Tue, 20 May 2014 07:01:53 +0000 (15:01 +0800)]
ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
The patch adds the RL6231 class device shared support for RT5640, RT5645 and
RT5651. The function of the DMIC clock calculation can be shared by RL6231
shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Xiubo Li [Mon, 19 May 2014 08:32:09 +0000 (16:32 +0800)]
ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
Since we cannot make sure the 'reg_size' will always be none zero here,
and then if 'reg_size' equals to zero, the kzalloc() will return ZERO_SIZE_PTR,
which equals to ((void *)16).
So this patch fix this with just doing the 'reg_size' zero check before calling
kzalloc().
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC: Add helper functions to cast from DAPM context to CODEC/platform
This is useful if we have a pointer to a DAPM context and know that it is a
CODEC or platform DAPM context and want to get a pointer to the CODEC or
platform.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Alexander Shiyan [Fri, 23 May 2014 11:11:01 +0000 (15:11 +0400)]
ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
Eukrea-i.MX51 board was converted to use DT, ie we no longer have a
MACH_EUKREA_MBIMXSD51_BASEBOARD symbol.
Transformation of other boards planned for the near future, so this
patch removes all these dependencies and restricts build of this
driver to ARCH_MXC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru> Signed-off-by: Mark Brown <broonie@linaro.org>
Markus Pargmann [Tue, 27 May 2014 08:24:23 +0000 (10:24 +0200)]
ASoC: fsl-ssi: Fix baudclock handling
The baudclock may be used and set by different streams.
Allow only the first stream to set the bitclock rate. Other streams have
to try to get to the correct rate without modifying the bitclock rate
using the SSI internal clock modifiers.
The variable baudclk_streams is introduced to keep track of the active
streams that are using the baudclock. This way we know if the baudclock
may be set and whether we may enable/disable the clock.
baudclock enable/disable is moved to hw_params()/hw_free(). This way we can
keep track of the baudclock in those two functions and avoid a running
clock while it is not used. As hw_params()/hw_free() may be called
multiple times for the same stream, we have to use baudclk_streams
variable to know whether we may enable/disable the clock.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Sascha Hauer [Tue, 27 May 2014 08:24:22 +0000 (10:24 +0200)]
ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode
In i2s master mode the fsl_ssi driver depends on someone calling
.set_tdm_slot correctly. In this mode though only a DC value of
2 is allowed, so set it in this case and no longer depend on
.set_tdm_slot.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Sascha Hauer [Tue, 27 May 2014 08:24:21 +0000 (10:24 +0200)]
ASoC: fsl-ssi: remove unnecessary spinlock
The baudclock_locked variable is only used in functions which
are serialized anyway from the core. No need to have a lock
around the variable, so remove it.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Sascha Hauer [Tue, 27 May 2014 08:24:20 +0000 (10:24 +0200)]
ASoC: fsl-ssi: set bitclock in master mode from hw_params
The fsl_ssi driver uses the .set_sysclk callback to configure the
bitclock for master mode. This is unnecessary since the bitclock
is known in hw_params. This patch configures the bitclock from .hw_params.
.set_dai_sysclk now sets a bitclock frequency which is preferred over
the default calculated bitclock frequency.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Markus Pargmann [Tue, 27 May 2014 08:24:19 +0000 (10:24 +0200)]
ASoC: fsl-ssi: make fsl,mode property optional
The simple soundcard binding has its own way for specifying the dai
format. To be able to use this binding we have to make the fsl,mode
property optional. As the property is used in existing devicetrees
keep the option around for compatibility reasons.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Sascha Hauer [Tue, 27 May 2014 08:24:18 +0000 (10:24 +0200)]
ASoC: fsl-ssi: introduce SoC specific data
Introduce a SoC data struct which contains the differences between
the different SoCs this driver supports. This makes it easy to support
more differences without having to introduce a new switch/case each
time.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Tested-By: Michael Grzeschik <mgr@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Jarkko Nikula [Wed, 28 May 2014 09:35:39 +0000 (12:35 +0300)]
ASoC: Intel: Use devm_snd_soc_register_card
Simplify byt-rt5640.c and haswell.c machine drivers by using
devm_snd_soc_register_card(). Remove also needless dev_set_drvdata()
from byt_rt5640_probe() since snd_soc_register_card() does it too.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Andy Shevchenko [Mon, 26 May 2014 13:56:31 +0000 (16:56 +0300)]
ASoC: Intel: remove duplicate headers
A few files contain duplicate headers. This patch removes the second entry of
duplicate in each file under question.
There is no functional changes.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Stored DSP DMA pointer must be cleared before starting the stream since
PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is
updated. In that case last position of previous stream was wronly returned.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Axel Lin [Fri, 23 May 2014 05:05:31 +0000 (13:05 +0800)]
ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
The new value argument needs proper shift to match the mask bit fields.
Signed-off-by: Axel Lin <axel.lin@ingics.com> Tested-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Imre Deak [Fri, 30 May 2014 07:52:29 +0000 (10:52 +0300)]
ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path
Baytrail and Haswell SST IPC don't stop the kernel thread in error and
cleanup path thus leaving orphan kernel thread behind in such a case.
Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free
hsw->msg.
[Jarkko: I edited the commit log a little] Signed-off-by: Imre Deak <imre.deak@intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec.
Jack detect code from Kevin Strasser <kevin.strasser@intel.com>, GPIO
resolving from Mika Westerberg <mika.westerberg@linux.intel.com> and fixes
and cleanups from Liam Girdwood <liam.r.girdwood@linux.intel.com>.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
Peter Ujfalusi [Fri, 30 May 2014 13:47:41 +0000 (16:47 +0300)]
ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function
For some unknown reason the parameters for snd_soc_test_bits() were in wrong
order:
It was:
snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */
while it should be:
snd_soc_test_bits(codec, reg, mask, val);
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
Takashi Iwai [Fri, 30 May 2014 08:15:58 +0000 (10:15 +0200)]
ALSA: firewire: Fix dependency on PCM and rawmidi
Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need
to give a proper dependency. For fixing and simplification, move the
selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section.
Then each driver doesn't have to select them but only
SND_FIREWIRE_LIB.
Reported-by: Jim Davis <jim.epost@gmail.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 23 May 2014 07:21:06 +0000 (09:21 +0200)]
ALSA: hda/analog - Fix silent output on ASUS A8JN
ASUS A8JN with AD1986A codec seems following the normal EAPD in the
normal order (0 = off, 1 = on) unlike other machines with AD1986A.
Apply the workaround used for Toshiba laptop that showed the same
problem.
Paul Bolle [Thu, 29 May 2014 16:58:25 +0000 (18:58 +0200)]
ALSA: gus: remove checks for CONFIG_SND_DEBUG_ROM
Checks for CONFIG_SND_DEBUG_ROM were added in v2.5.5 but a Kconfig
symbol SND_DEBUG_ROM was never added. These checks have always
evaluated to false. Remove them and the printk()s they hide.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Paul Bolle [Thu, 29 May 2014 16:51:44 +0000 (18:51 +0200)]
sound: remove checks for CONFIG_BCM_CS4297A_CSWARM
Checks for CONFIG_BCM_CS4297A_CSWARM were added in v2.6.11. The related
Kconfig symbol was never added so these checks always evaluated to true.
Remove them.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel Matuschek [Thu, 29 May 2014 14:08:03 +0000 (15:08 +0100)]
ASoC: wm8804: Allow control of master clock divider in PLL generation
WM8804 can run with PLL frequencies of 256xfs and 128xfs for
most sample rates. At 192kHz only 128xfs is supported. The
existing driver selects 128xfs automatically for some lower
samples rates. By using an additional mclk_div divider, it
is now possible to control the behaviour. This allows using
256xfs PLL frequency on all sample rates up to 96kHz. It
should allow lower jitter and better signal quality. The
behavior has to be controlled by the sound card driver,
because some sample frequency share the same setting. e.g.
192kHz and 96kHz use 24.576MHz master clock. The only
difference is the MCLK divider.
Signed-off-by: Daniel Matuschek <daniel@matuschek.net> Tested-by: Florian Meier <florian.meier@koalo.de> Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1381 and ADAU1781 audio
CODECs. The device is a low-power, 24-bit stereo audio CODEC with multiple
analog inputs and outputs, two digital microphone inputs and an I2S interface.
The device can be controlled either using I2C or SPI. The main difference
between the two variants is that the ADAU1781 has a freely programmable SigmaDSP
processor, while the ADAU1381 has a fixed function wind noise reduction filter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1361 and ADAU1761 CODECs.
The device is a a low-power, 24-bit stereo audio CODEC with multiple analog
input and outputs, one digital microphone input and an I2S interface. The device
can be controlled either via I2C or SPI. The main difference between the two
variants is that the ADAU1761 has a built-in SigmaDSP, while the ADAU1361 has
not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC: Add ADAU1X61 and ADAU1X81 CODECs common code
The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite
different in the analog domain. This patch adds support for the common parts of
the ADAU1X61 and ADAU1X81 CODECs.
The patch also restores some of the alphabetical order in the Makfile and
Kconfig.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
Takashi Sakamoto [Tue, 27 May 2014 15:14:47 +0000 (00:14 +0900)]
ALSA: bebob: Improve comments about stream format
Currently bebob driver apply Raw Audio Data channel (in IEC 61883-1:2002,
Multi Bit Linear Audio Data channel in IEC 61883-6:20005) to IEC 60958
Conformant Data channel because both fireworks and bebob based devices
can handle it by ignoring each label.
Takashi Sakamoto [Tue, 27 May 2014 15:14:46 +0000 (00:14 +0900)]
ALSA: bebob: Remove meaningless mutex_unlock()
Currently mutex_unlock() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
Takashi Sakamoto [Tue, 27 May 2014 15:14:45 +0000 (00:14 +0900)]
ALSA: bebob: Add static specifier to identifier with file scope
Some variables were declared without static even if they're not referred
to from external files. This commit make them local symbols for better
information-hiding by file unit.
Takashi Sakamoto [Tue, 27 May 2014 15:14:42 +0000 (00:14 +0900)]
ALSA: fireworks/bebob: Add suffix for long long integer literal
This commit adds suffix to register values of each device, to supress 'sparse'
warnings. Additionally, this commit changes offset values with integer literal.
Takashi Sakamoto [Tue, 27 May 2014 15:14:41 +0000 (00:14 +0900)]
ALSA: fireworks/bebob: Use the same type of variables as function parameters
The second argument of snd_efw_command_get_sampling_rate() means sampling
rate and its type is 'unsigned int'. But 'int' variable is passed as parameter.
It's better to apply the same type for the variable as its argument.
Similally, the type of variable for snd_efw_command_get_clock_source() and
avc_bridgeco_get_plug_type() should be the same type as each argument.
Takashi Sakamoto [Tue, 27 May 2014 15:14:40 +0000 (00:14 +0900)]
ALSA: fireworks/bebob: Change type of argument for sampling rate
Originally, I intent to this argument given with 'struct snd_pcm_runtime.rate'
or params_rate(). They return value of 'unsigned int'. So 'unsigned int' is
better for the type of this argument.
Takashi Sakamoto [Tue, 27 May 2014 15:14:39 +0000 (00:14 +0900)]
ALSA: fireworks: Use the same prototype for functions as actual declaration
There are two modes for Fireworks, IEC 61883 compliant or Windows.
So it's better to use enum type instead of int to express the intension,
even if C language specification defines to handle enum variables as usual
integer.
Takashi Sakamoto [Tue, 27 May 2014 15:14:36 +0000 (00:14 +0900)]
ALSA: firewire-lib: Use ARRAY_SIZE() instead of sizeof() for correct loop limit
This commit fixes a big for loop count with array. The limitation of loop
count should be calcurated with the number of elements in the array, not
with the number of bytes.
Aditionally, this commit apply the same declaration as a prototype in header
for the array.
projects / mirror_ubuntu-eoan-kernel.git / log