aboutsummaryrefslogtreecommitdiff
path: root/sound
AgeCommit message (Collapse)AuthorFilesLines
2011-03-14ALSA: ctxfi - Fix SPDIF status retrievalPrzemyslaw Bruski1-14/+5
SDPIF status retrieval always returned the default settings instead of the actual ones. Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-14ALSA: ctxfi - Fix incorrect SPDIF status bit maskPrzemyslaw Bruski1-1/+1
SPDIF status mask creation was incorrect. Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-14ALSA: ctxfi - Fix microphone boost codes/commentsPrzemyslaw Bruski1-10/+18
microphone boost was set at +12dB, not +20dB (like in Windows driver and in adc_conf structure declaration), some comments added. Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-13Merge branch 'fix/asoc' of ↵Linus Torvalds4-9/+19
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ASoC: Ensure WM8958 gets all WM8994 late revision widgets ASoC: Fix typo in late revision WM8994 DAC2R name ASoC: Use the correct DAPM context when cleaning up final widget set ASoC: Fix broken bitfield definitions in WM8978 ASoC: AM3517: Update codec name after multi-component update
2011-03-12Merge branch 'master' of ↵Russell King26-53/+310
git://git.kernel.org/pub/scm/linux/kernel/git/tglx/linux-2.6-tcc into devel-stable
2011-03-11ALSA: atiixp - Fix wrong time-out checks during ac-link resetTakashi Iwai2-2/+2
The time-out in snd_atiixp_aclink_reset() is wrongly checked, and it resulted in exiting from the loop at the first iteration. Reported-by: Amir Shamsuddin <AmirS2+alsa@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11ALSA: intel8x0m: append 'm' to "r_intel8x0"Paul Bolle1-3/+3
Appending an 'm' will distinguish it from a similar struct in intel8x0.c Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11ALSA: intel8x0m: add 'm' as "suffix" to static functionsPaul Bolle1-49/+49
Adding an 'm' will distinguish them from identical names in intel8x0.c. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11ALSA: intel8x0m: wait a bit before warm reset checkPaul Bolle1-1/+2
At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. A way to fix this is to call usleep_range() after writing to the register involved. A handful of tests suggest 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11ALSA: usbaudio: implement USB autosuspendOliver Neukum6-22/+118
Devices are autosuspended if no pcm nor midi channel is open Mixer devices may be opened. This way they are active when in use to play or record sound, but can be suspended while users have a mixer application running. [Small clean-ups using static inline by tiwai] Signed-off-by: Oliver Neukum <oneukum@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11ALSA: usbaudio: fix suspend/resumeOliver Neukum3-3/+34
- ESHUTDOWN must be correctly handled - the optional interrupt endpoint's URB must be stopped and restarted Signed-off-by: Oliver Neukum <oneukum@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11Merge branch 'fix/misc' into topic/miscTakashi Iwai45-172/+347
2011-03-11ASoC: mini2440: Fix uda134x codec problem.Marek Belisko2-2/+4
ASoC audio for mini2440 platform in current kenrel doesn't work. First problem is samsung_asoc_dma device is missing in initialization. Next problem is with codec. Codec is initialized but never probed because no platform_device exist for codec driver. It leads to errors during codec binding to asoc dai. Next problem was platform data which was passed from board to asoc main driver but not passed to codec when called codec_soc_probe(). Following patch should fix issues. But not sure if in correct way. Please review. Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
2011-03-11ASoC: Fix spacing in MAX8950Mark Brown1-1/+1
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-11ASoC: PXA: Z2: Fix codec pin nameVasily Khoruzhick1-1/+1
MONO was renamed to MONO1. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
2011-03-11ASoC: PXA: z2: Mute internal speaker when headphones are connectedVasily Khoruzhick1-0/+5
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-11ASoC: Add MAX9850 codec driverChristian Glindkamp4-0/+433
This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-11ALSA: Add snd_ctl_activate_id()Takashi Iwai1-0/+46
Added a new API function snd_ctl_activate_id() for activate / inactivate the control element dynamically. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-10Merge branch 'for_2.6.39/pm-misc' of ↵Tony Lindgren5-15/+57
ssh://master.kernel.org/pub/scm/linux/kernel/git/khilman/linux-omap-pm into omap-for-linus
2011-03-11Merge branch 'next-exynos4' into for-nextKukjin Kim1-1/+1
2011-03-10ALSA: hda: Prevent writing ICH6_PCIREG_TCSEL on AMD systemsAdam Lackorzynski1-2/+5
azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL although this looks to be only defined on Intel systems and has a different meaning on AMD systems. On AMD systems the PCI interrupt pin control register is modified instead. Since the meaning of offset 0x44 in device specific configuration space is unknown for devices by other vendors, we only exclude AMD systems to retain the current behaviour. Signed-off-by: Adam Lackorzynski <adam@os.inf.tu-dresden.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: HDA: Realtek ALC88x: Do not over-initialize speakers and hp that are ↵David Henningsson1-16/+21
primary outputs Do not initialize again the what has already been initialized as multi outs, as this breaks surround speakers. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88xDavid Henningsson1-3/+6
Without this change, a volume control named "Surround" or "Side" would get an unnecessary index, causing it to be ignored by the vmaster and PulseAudio. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: HDA: Fix volume control naming for surround speakers on Realtek ↵David Henningsson1-1/+3
auto-parser When more than one pair of internal speakers is present, allow names according to their channels. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: HDA: Enable surround and subwoofer on Lenovo Ideapad Y530David Henningsson1-0/+10
The pin config values would change the association instead of the sequence, this commit fixes that up. Tested-by: Bartłomiej Żogała <nusch88@gmail.com> Cc: <stable@kernel.org> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: usb-audio - Add "cval->res = 384" quirk for Logitech Webcam C600Alexey Fisher1-0/+1
One more affected devices: Logitech Webcam C600 (046d:0808) Volume range before quirk is 6400, after (also real) is 16. Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10ALSA: usb-audio - Add volume range check and warn if it too bigAlexey Fisher1-0/+16
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10Merge branch 'for-2.6.38' of ↵Takashi Iwai3-8/+18
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
2011-03-10Merge branch 'for-2.6.38' into for-2.6.39Mark Brown1-1/+9
2011-03-10Merge branch 'for-rmk' of git://git.pengutronix.de/git/imx/linux-2.6 into ↵Russell King2-2/+4
devel-stable
2011-03-09ASoC: Ensure WM8958 gets all WM8994 late revision widgetsMark Brown1-0/+8
Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
2011-03-09ASoC: Fix typo in late revision WM8994 DAC2R nameMark Brown1-1/+1
Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
2011-03-09sound: Use sound_register_*() for additional OSS minor devicesTakashi Iwai2-36/+23
Since OSS driver creates the device entries for /dev/audio* and /dev/dspW* by itself without coping with sound_core, it leads to conflicts with others and let sysfs spewing warnings. This patch rewrites the registration part of OSS driver to use the standard method also for additional minor devices. Reported-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl) Tested-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl) Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-09ASoC: Enable 192kHz sample rate for EP93xx.Alexander Sverdlin2-4/+4
Enable 192kHz sample rate for EP93xx. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-09ASoC: Improve EP93xx I2S clocks management.Alexander Sverdlin1-10/+9
Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function (by selecting SCLK/LRCLK=64 where possible). All other rates should work as before. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-09ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.Alexander Sverdlin1-8/+8
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-09ASoC: Manage mode and rate bits correctly for CS4271 CODEC.Alexander Sverdlin1-30/+53
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-09Merge branch 'for-2.6.38' into for-2.6.39Mark Brown1-1/+1
Conflicts: sound/soc/codecs/wm8978.c sound/soc/soc-dapm.c
2011-03-09ASoC: Fix double addition of prefixes due to widget prefixingMark Brown1-3/+26
We're not only prefixing all controls, we're also prefixing the widget names in the runtime data. This causes us to add the prefix twice - once when using the widget name to generate the control name and once when adding the control. Really we shouldn't be prefixing the widget names at all, the matching code should be handing this as we always know which DAPM context a widget came from and always display the widget name in terms of a DAPM context. However, we're quite close to the merge window and that's relatively invasive. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Reported-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-09ASoC: Use the correct DAPM context when cleaning up final widget setMark Brown1-1/+1
Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-09ASoC: Fix broken bitfield definitions in WM8978Mark Brown1-6/+8
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
2011-03-09ASoC: Simplify WM9081 speaker startup by using widgets for spaker outputMark Brown1-26/+6
Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-09ASoC: AM3517: Update codec name after multi-component updateAbhilash K V1-1/+1
The i2c client device name (".2-001a" in this case, including the separator period) for the AIC23 codec on the TI AM3517-EVM was appended to the codec_name member of am3517evm_dai to resolve the names mismatch happening in soc_bind_dai_link(), due to which the card was not getting registered. Signed-off-by: Abhilash K V <abhilash.kv@ti.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-09ASoC: omap: rx51: Enable McBSP2 sidetoneJarkko Nikula1-0/+4
McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-08ASoC: sgtl5000: use after free in ldo_regulator_register()Dan Carpenter1-1/+3
The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter <error27@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-03-08ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()Mark Brown2-16/+45
Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-08ASoC: Warn rather than set a silly constraint when we can't do symmetryMark Brown1-0/+10
Symmetric rate configuration can fail if the second stream starting tries to apply the symmetric constraint before the first stream has got far enough to pick a rate. Rather than try to enforce a nonsensical rate of 0Hz log a warning and allow the application to carry on. Things might go wrong later on but the user will know about it and there's unlikely to be lasting damage. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-08ASoC: Refactor symmetric_rates check to reduce indentationMark Brown1-14/+14
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-08ASoC: Convert WM9081 SYSCLK configuration to be device wideMark Brown1-3/+5
Also respace the CODEC ops a bit for legibility. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-08ASoC: Provide CODEC clocking operations and API callsMark Brown1-0/+46
When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Powered by cgit, Gruvboxed by Blaster4385.