- 30 10月, 2009 1 次提交
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由 Roel Kluin 提交于
The variables are unsigned so the test `>= 0' is always true, the `< 0' test always fails. In these cases the other part of the test catches wrapped values. In dac_audio_write() there does not occur a test for wrapped values, but the test appears redundant. Signed-off-by: NRoel Kluin <roel.kluin@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 10月, 2009 1 次提交
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由 Tobias Hansen 提交于
snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd This is the correct error number for telling the USB system that this driver is not for the device. Signed-off-by: NTobias Hansen <Tobias.Hansen@physik.uni-hamburg.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 10月, 2009 3 次提交
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由 Wu Zhangjin 提交于
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family machines, and checked it with ARCH=x86_64, no relative compiling warnings & errors, so, remove the platform dependency directly. Reported-by: rixed@happyleptic.org Acked-by: NAndres Salomon <dilinger@collabora.co.uk> Signed-off-by: NWu Zhangjin <wuzhangjin@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Krzysztof Helt 提交于
The chipsets with the isa_dma_bridge_buggy set do not stop DMA during DMA counter reads. The DMA counter is read in two 8-bit read steps on x86 platform. Sometimes, such reads happen during higher byte change so the lower byte is already decremented (rolled over) but the higher byte is not. It introduces an error that position is moved 256 bytes ahead of the true position. Thus, the next DMA position read can return a lower value then the previous read. If the DMA position is decreased (reversed) the ALSA subsystem is tricked into the playback underrun error and resets the playback. It results in a "pop" during a playback. Work around the issue by reading the counter twice and choosing a higher value. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Krzysztof Helt 提交于
The C4231 control set is a superset of the AD1848 control set so reuse the CS4231 controls definitions for the AD1848. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 10月, 2009 2 次提交
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由 Krzysztof Helt 提交于
Convert CS4231 mixer to dB scale after AD1848 mixer. Also, add missing microphone boost control for the AD1848 and correct wrong bits for loopback volume on the AD1848. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Krzysztof Helt 提交于
Fix coding style errors in the driver. Also, add missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 10月, 2009 2 次提交
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由 Takashi Iwai 提交于
Module parameters shouldn't be marked as __devinitdata since they can be referred via sysfs even after probing. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Krzysztof Helt 提交于
Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 10月, 2009 1 次提交
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由 Krzysztof Helt 提交于
There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 10月, 2009 1 次提交
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由 Krzysztof Helt 提交于
The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 28 9月, 2009 1 次提交
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由 Krzysztof Helt 提交于
Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: NKrzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 22 9月, 2009 1 次提交
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由 Tobias Hansen 提交于
Adds support for US-144 when attached on USB1.1. Unlike the US-122L it uses both USB interfaces 0 and 1. Signed-off-by: NTobias Hansen <Tobias.Hansen@physik.uni-hamburg.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 9月, 2009 1 次提交
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由 Takashi Iwai 提交于
The headphone and speaker mixer elements aren't properly set for MSI GX620 with targa-8ch-dig quirk. Also fixed the speaker volume control for other ALC883-targa quirks, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 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 3 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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由 Takashi Iwai 提交于
The pin setup for Dell S14 quirk is rather wrong for the latest driver. Fixed pin 0x0a, 0x0b, 0x0d and 0x0f. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Remove unnecessary (and buggy) init sequences left for IDT92HD83* codecs in the previous fixes. The DACs are now dynamically connected, thus shouldn't be set statically in init verbs. Also, the mono_nid is detected dynamically, thus shouldn't be set staticaly, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 9月, 2009 5 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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由 Takashi Iwai 提交于
Add the quirk entry for HP dv6. Also add a workaround for the headphone detection by setting hp_detect=1 beforehand. Without this, the driver won't do auto-muting because BIOS doesn't give any HP pin but only a line-out pin. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
It's possible that hp_detect is set even though no headphone pin is detected. The driver issues, however, an unsol event only to hp_pins[0], which can be invalid. This patch adds the check of the valid pin to send an unsol event at initialization and resume callbacks. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
A smiliar fix for IDT 92HD71Bxx codecs like the previous commit for other IDT/STAC codecs. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines. However, currently we don't set it unless the model is specified just for safety reason. But, most machines do need this bit, so this safety handling is rather annoying. This patch enables GPIO0 setup as default for them. Many HP / Dell laptops should work even without model override with this change. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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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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- 11 9月, 2009 1 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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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 1 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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