- 17 9月, 2009 1 次提交
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由 Huang Weiyi 提交于
Remove unused #include <linux/version.h>('s) in sound/soc/codecs/ad1836.c sound/soc/codecs/ad1938.c sound/soc/codecs/wm8974.c Signed-off-by: NHuang Weiyi <weiyi.huang@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 15 9月, 2009 1 次提交
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由 Jassi 提交于
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK is dead due to improper initialization of CPU or CODEC, the system gets stuck in the loop because jiffies may never get updated. Implemented counter based wait mechanism for atleast the same timeout period. Signed-off-by: NJassi <jassi.brar@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 14 9月, 2009 1 次提交
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由 Mark Brown 提交于
Also display streams all the time while we're here. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 12 9月, 2009 1 次提交
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由 Julia Lawall 提交于
Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: NJulia Lawall <julia@diku.dk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 09 9月, 2009 2 次提交
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由 Manuel Lauss 提交于
This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: NManuel Lauss <manuel.lauss@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Takashi Iwai 提交于
Increase the limit of PCM substreams to 128. The default value is unchanged; only the max accept value is increased. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 08 9月, 2009 6 次提交
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由 Takashi Iwai 提交于
Added the debug proc file to see or change the snd_pcm_hardware fields to emulate. The parameters can be changed by writing to a proc file like: # echo periods_min 4 > /proc/asound/card1/dummy_pcm Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Add appropriate const prefix to char * arguments in proc helper functions. Also fixed the caller side to be proper const pointers. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Re-export snd_pcm_format_name() function to be used outside the PCM core. As a first example, usbaudio is changed to use it now again. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The HP laptops with ALC268 codec seem working better with model=auto than model=toshiba; e.g. the auto model fixes missing digital outputs. Let's fix quirk entry to choose auto model explicitly. Tested-by: NJens Jorgensen <jbj1@ultraemail.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Sophie Hamilton 提交于
Fix minimum period size for cs46xx cards. This fixes a problem in the case where neither a period size nor a buffer size is passed to ALSA; this is the case in Audacious, OpenAL, and others. Signed-off-by: NSophie Hamilton <kernel@theblob.org> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Mark Brown 提交于
It's the 8th enum of a zero indexed array. This is why I don't let new drivers use these arrays of enums... Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
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- 07 9月, 2009 4 次提交
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由 Takashi Iwai 提交于
The struct snd_monitor_file is used locally only in sound/core/init.c, thus it should be moved there from the public sound/core.h. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
When the volume is changed continuously (e.g., when the user drags a volume slider with the mouse), the driver does lots of I2C writes. Apparently, the sound chip can get confused when we poll the I2C status register too much, and fails to complete a read from it. On the PCI-E models, the PCI-E/PCI bridge gets upset by this and generates a machine check exception. To avoid this, this patch replaces the polling with an unconditional wait that is guaranteed to be long enough. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Tested-by: Johann Messner <johann.messner at jku.at> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Instead of allocating the real buffers, use a fake buffer and ignore read/write in the dummy driver so that we can save the resources. For mmap, a single page (unique to the direction, though) is reused to all buffers. When the app requires to read/write the real buffers, pass fake_buffer=0 module option at loading time. This will get back to the old behavior. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 ddiaz@cenditel.gob.ve 提交于
The model clevo-m540r was created with 6-channel and digital support. All functions verified except spdif. Tested with a VIT D2000 laptop which has: [lspci extract] Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: CLEVO/KAPOK Computer Device [1558:5409] [/proc/asound/card0/codec\#0 header] Codec: Realtek ALC883 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0883 Subsystem Id: 0x15585409 Revision Id: 0x100002 [Added a comment about HP mute and the model description by tiwai] Signed-off-by: NDhionel Diaz <ddiaz@cenditel.gob.ve> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 9月, 2009 1 次提交
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由 Robert Schwebel 提交于
Today's linux-next fails to build with sound/arm/pxa2xx-ac97.c: In function 'pxa2xx_ac97_probe': sound/arm/pxa2xx-ac97.c:211: error: 'pxa2xx_audio_ops_t' has no member named 'codec_data' make[2]: *** [sound/arm/pxa2xx-ac97.o] Error 1 It looks like commit e2365bf3 has introduced this; patch below. Signed-off-by: NRobert Schwebel <r.schwebel@pengutronix.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 04 9月, 2009 1 次提交
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由 Takashi Iwai 提交于
Fix the expire-time calculation in the systimer mode when the buffer size isn't aligned to the period size. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 9月, 2009 4 次提交
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由 Takashi Iwai 提交于
In the system-timer mode, snd-dummy driver issues each tick to update the position. This is highly inefficient and even inaccurate if the timer can't be triggered at each tick. Now rewritten to wake up only at the period boundary. The position is calculated from the current jiffies. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Allow snd-dummy driver to use high-res timer as its timing source instead of the system timer. The new module option "hrtimer" is added to turn on/off the high-res timer support. It can be switched even dynamically via sysfs. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The card model detection code introduced in 2.6.30 that tries to work around partially broken EEPROM contents by reading the EEPROM directly does not handle cards where the EEPROM has been omitted. In this case, we have to use the default ID to allow the driver to load. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Reported-and-tested-by: NOzan Çağlayan <ozan@pardus.org.tr> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Mark Brown 提交于
Avoids potential issues if we read back unexpected values during a read/modify/write cycle. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 02 9月, 2009 4 次提交
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由 Takashi Iwai 提交于
Added the quirk for Alienware M17x with IDT 92HD73* codec chip. It has two HP and one line-out jack, one mic jack, a built-in speaker and a built-in mic. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Due to the previous fix of input source for IDT92HD73xx, the amp mux and amp vol stuff became unused. Let's rip off dead codes. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Fix the mux_nids to select directly the input source instead of mux mixers so that it works with the current mux enum handler for IDT 92HD73xx codecs. Also, clean up useless / unnecessary mixer controls and init verbs. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Fix the old dead CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 9月, 2009 5 次提交
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由 jassi brar 提交于
Bug was caught while trying to use WM8580 as I2S master on SMDK. Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved by referring to WM8580A manual and setting mask value correctly and making the code to not touch 'reserved' bits of PLL4 register. Signed-off-by: NJassi <jassi.brar@samsung.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Barry Song 提交于
As discussed, the patch uses the original TDM order without rewriting. For the match between TDM slot number and audio channel number, a new API need be added. Signed-off-by: NBarry Song <21cnbao@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Takashi Iwai 提交于
Unmute the docking-station line-out as default on machines with AD1984A codec chip. It can be still muted via "Dock" mixer switch. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Added another entry for Nvidia HDMI device (10de:0003). Reference: kernel bug#14097 http://bugzilla.kernel.org/show_bug.cgi?id=14097Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
Do not forget to program the MCLK ratio for the I2S output. Otherwise, the master clock frequency can be too high for the DACs at sample frequencies above 96 kHz. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 31 8月, 2009 6 次提交
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由 Roel Kluin 提交于
Allocation may fail, show if it did. Signed-off-by: NRoel Kluin <roel.kluin@gmail.com> [Additional fix for invalid runtime->oss.prepare flag set by tiwai] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
A similar initialization of GPIO1 pin like mobile model is needed for laptop model, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Add the support of automatic mute and mic-switching of the docking station HP and mic plugs for AD1984A laptop model for some HP machines. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Fix the headphone pin routing of ALC268/ALC269 codecs. Using alc882 routine doesn't work because alc268/alc269 parser assumes the independent DACs for both HP and speaker outputs. Need to assign the DAC depending on the pin. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1 quirk with ALC889A codec, which caused the silent headphone output. Also, this gives the individual Headphone and Speaker volume controls. Reference: kernel bug#14078 http://bugzilla.kernel.org/show_bug.cgi?id=14078Signed-off-by: NTakashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
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由 Takashi Iwai 提交于
In patch_vt1708(), the check of MUX nids is missing and this results in the -EINVAL error in accessing Input Source mixer element. Simpliy adding the call of get_mux_nids() fixes the problem. Reference: Novell bnc#534904 https://bugzilla.novell.com/show_bug.cgi?id=534904Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 8月, 2009 2 次提交
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由 Takashi Iwai 提交于
So far, the digital mic capture volume wasn't created. This is because IDT codecs have output amps for digital mics, not input amps, while input amps should be used for other analog pins. Thus the automatic capture volume creation should check both directions for digital mics. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Jarkko Nikula 提交于
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have specifications to check it) have additional CLKR and FSR pins for McBSP1 receiver. Reset default is that receiver is using bit clock and frame sync signal from those pins but it is possible to configure to use also CLKX and FSX pins as well. In fact, other McBSP ports are doing that internally that transmitter and receiver share the CLKX and FSX. Add functionaly that machine drivers can set the CLKR and FSR sources by using the snd_soc_dai_set_sysclk. Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue. Signed-off-by: NJarkko Nikula <jhnikula@gmail.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 28 8月, 2009 1 次提交
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由 Chaithrika U S 提交于
Also, the codec setup data structure has to remain for successful probe. Signed-off-by: NChaithrika U S <chaithrika@ti.com> Signed-off-by: NKevin Hilman <khilman@deeprootsystems.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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