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* ALSA: pcxhr: hard dependency on the standard fw loaderTakashi Iwai2012-11-222-86/+1
| | | | | | | Like the previous commit for mixart, drop the home-baked fw loader code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: mixart: hard dependency on the standard fw loaderTakashi Iwai2012-11-222-76/+1
| | | | | | | | | | It makes no longer sense to keep the old hwdep user-space firmware loading, which has been deprecated since ages ago. Just add a hard dependency on CONFIG_FW_LOADER and drop the useless code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Don't overwrite the pin default configsTakashi Iwai2012-11-221-42/+3
| | | | | | | | | | | | | | | | | | Since we keep the pin default config values anyway internally, we don't have to set the values in the codec. This patch removes the code writing the pincfg values. As a gratis bonus, we can remove also the code restoring the original pincfg values at PM resume or module free. This will give us more benefit, as it can reduce the unnecessary power-up of codecs. This won't change the driver functionality. The only difference would be that the codec proc file will show the original pincfg values instead of the actually referred values. The actually referred values can be determined from sysfs *_pin_configs files. (Also hda-emu was updated to follow this change.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Remove shutup calls in free callbacksTakashi Iwai2012-11-213-4/+0
| | | | | | | | The free callback is called at the state where no extra verbs are executed, thus calling *_shutup() is useless, as it's checking the shutdown flag. Remove such superfluous calls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Use standard helper functions in patch_hdmi.cTakashi Iwai2012-11-211-4/+3
| | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: process pending stop at PCM hw_free and closeTakashi Iwai2012-11-211-2/+2
| | | | | | | | PCM hw_free and close should wait until all the pending stop operations have been finished. Basically only PCM trigger callback should use non-wait calls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: stop both data and sync endpoints asynchronouslyTakashi Iwai2012-11-213-9/+13
| | | | | | | | | | | | | | As we are stopping the endpoints asynchronously now, it's better to trigger the stop of both data and sync endpoints and wait for pending stopping operations, instead of the sequential trigger-and-wait procedure. So the wait argument in snd_usb_endpoint_stop() is dropped, and it's expected that the caller synchronizes explicitly by calling snd_usb_endpoint_sync_pending_stop(). (Actually there is only one place calling this, so it was safe to change.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: simplify endpoint deactivation codeTakashi Iwai2012-11-211-16/+7
| | | | | | | | For further code simplification, drop the conditional call for usb_kill_urb() with can_wait argument in deactivate_urbs(), and use only usb_unlink_urb() and wait_clear_urbs() pairs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: simplify snd_usb_endpoint_start/stop argumentsTakashi Iwai2012-11-213-26/+21
| | | | | | | | | Reduce the redundant arguments for snd_usb_endpoint_start() and snd_usb_endpoint_stop(). Also replaced from int to bool. No functional changes by this commit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Deprecate async_unlink optionTakashi Iwai2012-11-213-9/+1
| | | | | | | The async unlink behavior has been working over years. The option was provided only as a workaround for 2.4.x kernel. Let's get rid of it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Return meaningful error codes instead of -1 in format.cSachin Kamat2012-11-211-5/+5
| | | | | | | | | Also, silences the following smatch warning: sound/usb/format.c:170 parse_audio_format_rates_v1() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb/6fire: Fix potential NULL pointer dereference in comm.cSachin Kamat2012-11-211-1/+2
| | | | | | | | 'rt' was dereferenced before the NULL check. Moved the code after the check. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Fix return code in pcm_native.cSachin Kamat2012-11-211-1/+1
| | | | | | | | | | Return the value obtained from snd_pcm_hw_constraint_minmax() instead of -EINVAL. Silences the following smatch warning: sound/core/pcm_native.c:2003 snd_pcm_hw_constraints_complete() info: why not propagate 'err' from snd_pcm_hw_constraint_minmax() instead of -22? Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: PCM: Remove redundant null check before kfreeSachin Kamat2012-11-211-2/+1
| | | | | | | kfree on a null pointer is a no-op. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-1916-44/+629
|\ | | | | | | | | | | | | | | Merge the recent HD-audio codec change for fixing recursive suspend calls. Conflicts: sound/pci/hda/hda_codec.c
| * ALSA: hda - Limit runtime PM support only to known Intel chipsTakashi Iwai2012-11-191-19/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | We've got a report that the runtime PM may make the codec the unresponsive on AMD platforms. Since the feature has been tested only on the recent Intel platforms, it's safer to limit the support to such devices for now. This patch adds a new DCAPS bit flag indicating the runtime PM support, and mark it for Intel controllers. Reported-and-tested-by: Julian Wollrath <jwollrath@web.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix recursive suspend/resume callTakashi Iwai2012-11-192-2/+10
| | | | | | | | | | | | | | | | | | | | | | | | When the bus reset is performed during the suspend/resume (including the power-saving too), it calls snd_hda_suspend() and snd_hda_resume() again, and deadlocks eventually. For avoiding the recursive call, add a new flag indicating that the PM is being performed, and don't go to the bus reset mode when it's on. Reported-and-tested-by: Julian Wollrath <jwollrath@web.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'usb-midi-fix-3.7' of git://git.alsa-project.org/alsa-kprivate ↵Takashi Iwai2012-11-191-2/+6
| |\ | | | | | | | | | | | | | | | | | | into for-linus Merge a regression fix for USB MIDI on non-standard usb-audio drivers by Clemens.
| | * ALSA: ua101, usx2y: fix broken MIDI outputClemens Ladisch2012-11-181-2/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) added autosuspend code to all files making up the snd-usb-audio driver. However, midi.c is part of snd-usb-lib and is also used by other drivers, not all of which support autosuspend. Thus, calls to usb_autopm_get_interface() could fail, and this unexpected error would result in the MIDI output being completely unusable. Make it work by ignoring the error that is expected with drivers that do not support autosuspend. Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com> Reported-by: Devin Venable <venable.devin@gmail.com> Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk> Reported-by: Jannis Achstetter <jannis_achstetter@web.de> Reported-by: Rui Nuno Capela <rncbc@rncbc.org> Cc: Oliver Neukum <oliver@neukum.org> Cc: 2.6.39+ <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * | ALSA: usb-audio: Fix mutex deadlock at disconnectionTakashi Iwai2012-11-141-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent change for USB-audio disconnection race fixes introduced a mutex deadlock again. There is a circular dependency between chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a device is opened during the disconnection operation: A. snd_usb_audio_disconnect() -> card.c::register_mutex -> chip->shutdown_rwsem (write) -> snd_card_disconnect() -> pcm.c::register_mutex -> pcm->open_mutex B. snd_pcm_open() -> pcm->open_mutex -> snd_usb_pcm_open() -> chip->shutdown_rwsem (read) Since the chip->shutdown_rwsem protection in the case A is required only for turning on the chip->shutdown flag and it doesn't have to be taken for the whole operation, we can reduce its window in snd_usb_audio_disconnect(). Reported-by: Jiri Slaby <jslaby@suse.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: fm801: precedence bug in snd_fm801_tea575x_get_pins()Dan Carpenter2012-11-141-3/+8
| | | | | | | | | | | | | | | | | | | | | | | | There is a precedence bug because | has higher precedence than ?:. This code was cut and pasted and I fixed a similar bug a few days ago. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: es1968: precedence bug in snd_es1968_tea575x_get_pins()Dan Carpenter2012-11-131-3/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I don't think this works as intended. '|' higher precedence than ?: so the bitwize OR "0 | (val & STR_MOST)" is a no-op. I have re-written it to be more clear. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-11-138-12/+573
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 A few small fixes plus a large but simple change for WM5102 which writes out a bunch of register updates to the device when we enable the clock as recommended following chip evaluation.
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| | *---------. \ Merge branches 'fix/arizona', 'fix/core', 'fix/cs42l52', 'fix/mxs', ↵Mark Brown2012-11-138-12/+573
| | |\ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'fix/samsung' and 'fix/wm8978' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into tmp
| | | | | | | | * | ASoC: wm8978: pll incorrectly configured when codec is masterEric Millbrandt2012-11-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When MCLK is supplied externally and BCLK and LRC are configured as outputs (codec is master), the PLL values are only calculated correctly on the first transmission. On subsequent transmissions, at differenct sample rates, the wrong PLL values are used. Test for f_opclk instead of f_pllout to determine if the PLL values are needed. Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | | | | | * | | ASoC: bells: Correct type in sub speaker DAI name for WM5102Charles Keepax2012-11-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | | * | | ASoC: bells: Select WM1250-EV1 Springbank audio I/O moduleDimitris Papastamos2012-11-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Ensure we select the WM1250-EV1 as the current software system configuration demands it. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | | * | | ASoC: bells: Add missing select of WM0010Dimitris Papastamos2012-11-021-0/+1
| | | | | | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | * | | ASoC: mxs-saif: Fix channel swap for 24-bit formatFabio Estevam2012-11-021-4/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner <DWinner@tc-helicon.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Tested-by: Dan Winner <DWinner@tc-helicon.com> Acked-by: Dong Aisheng <dong.aisheng@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | * | | ASoC: mxs-saif: Add MODULE_ALIASFabio Estevam2012-11-011-0/+1
| | | | | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add MODULE_ALIAS information. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Acked-by: Dong Aisheng <dong.aisheng@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | * | | ASoC: cs42l52: fix the return value of cs42l52_set_fmt()Wei Yongjun2012-11-071-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the return value of cs42l52_set_fmt() when clock inversion is not allowed and also remove the useless variable ret. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) [We had been assigning to ret but then ignoring the value we assgined -- broonie] Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | | * | | | ASoC: core: Double control update err for snd_soc_put_volsw_sxMukund Navada2012-11-091-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_put_volsw_sx function fails to update second control if first control is updated by snd_soc_update_bits_locked. Signed-off-by: Mukund Navada <navada@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | | * | | | ASoC: dapm: Use card_list during DAPM shutdownMisael Lopez Cruz2012-11-091-1/+1
| | | | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | DAPM shutdown incorrectly uses "list" field of codec struct while iterating over probed components (codec_dev_list). "list" field refers to codecs registered in the system, "card_list" field is used for probed components. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | * | / / ASoC: wm5102: Write register value corrections after SYSCLK is enabledMark Brown2012-10-301-1/+551
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Evalation of the WM5102 has identified a number of register values which should be written after SYSCLK is enabled on revision A in order to improve performance. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | / | | ALSA: hda - Add a missing quirk entry for iMac 9,1Takashi Iwai2012-11-121-0/+1
| |/ / / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is another variant of iMac 9,1 with a different codec SSID. Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com> Cc: <stable@vger.kernel.org> [v3.3+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: hda - Fix Acer Aspire models with analog mics.Dylan Reid2012-11-191-0/+29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Acer Aspire AO756 has an analog built-in mic on nid 0x1b and an external mic on nid 0x19. The BIOS doesn't set this up. The mic detect on this Acer Aspire netbook and Acer C7 ChromeBook is only valid when the headphone is plugged. The detect circuit relies on the tip detect switch being closed on the jack. Tell hda_jack to ignore the mic sense unless the headphones are plugged. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: hda - Allow jack state to depend on another jackDylan Reid2012-11-192-9/+67
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Introduce the concept of a "gated" jack. The gated jack's pin sense is only valid when the "gating" jack is plugged. This requires checking the gating jack when the gated jack changes and re-checking the gated jack when the gating jack is plugged/unplugged. This allows handling of devices where the mic jack detect floats when the headphone jack is unplugged. [Rewritten for fixing the possible snd_array reallocation, covering the missing callback calls and jack sync operations, as well as some code cleanups -- tiwai] Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | sound: oss/sb_audio: cap value in sb201_audio_set_speed()Dan Carpenter2012-11-181-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We set "s" before we have capped "speed" so it could be the wrong value. This could lead to a divide by zero bug. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: usb-audio: use bitmap_weightJoe Perches2012-11-171-6/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use bitmap_weight to count the total number of bits set in bitmap. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: core: fix NULL checking in snd_pcm_plug_slave_size()Xi Wang2012-11-141-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The dereference snd_pcm_plug_stream(plug) should come after the NULL check snd_BUG_ON(!plug). Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: core: fix NULL checking in snd_pcm_plug_client_size()Xi Wang2012-11-141-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The dereference snd_pcm_plug_stream(plug) should come after the NULL check snd_BUG_ON(!plug). Signed-off-by: Xi Wang <xi.wang@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6Martin Schwenke2012-11-131-0/+52
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by: Martin Schwenke <martin@meltin.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'stanton-cs1-driver' of ↵Takashi Iwai2012-11-123-0/+542
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | git://git.alsa-project.org/alsa-kprivate into for-next
| * | | | | ALSA: firewire: add Stanton SCS.1d/1m driverClemens Ladisch2012-11-123-0/+542
| | |_|_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | Add a MIDI driver for the Stanton FireWire DJ controllers. Tested-by: Sean M. Pappalardo - D.J. Pegasus <spappalardo@mixxx.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
* | | | | ALSA: ice1724: Fix build error without CONFIG_PM_SLEEPTakashi Iwai2012-11-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bah, forgot this again... Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: ice1724: enable suspend on unknown ICE1724 cardsOndrej Zary2012-11-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Assume that unknown ICE1724-based cards are AC97-only that can suspend without any additional card-specific code. This fixes suspend on Gainward Hollywood@Home 7.1. Signed-off-by: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: hda - Removed unused non-standard name "C/LFE"David Henningsson2012-11-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | A closer look shows that the name is not even used and can be removed. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: hda - Apply a proper chmap for built-in 2.1 speakersTakashi Iwai2012-11-087-2/+40
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When 2.1 speakers are detected, use the corresponding channel map instead of the standard map with front+rear surrounds. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | ALSA: hda - Give standard "Bass Speaker" mixer for 2.1 speakersTakashi Iwai2012-11-084-11/+39
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When two built-in speakers are found on the machine, we can suppose it's rather a 2.1 speaker system with a bass output instead of front/surround channels. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'for-linus' into for-nextTakashi Iwai2012-11-0821-41/+98
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