summaryrefslogtreecommitdiff
path: root/sound
AgeCommit message (Collapse)Author
2010-02-25ALSA: hda - Add/fix ALC269 FSC and Quanta modelsKailang Yang
Specify proper quirk models for FSC and Quanta machines with ALC269 codec. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-25ALSA: hda - Add ALC670 codec supportKailang Yang
- Fixed alc_subsystem_id( ) typo and add new function. - !(ass & 0x100000)) ==> Delete this check. It is unnecessary check. - Add porti - ALC670 support Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-24ALSA: hda - remove unnecessary msleep on power state transitionsZhang, Rui
This will save ~15ms boot time. The first 10ms sleep was introduced in commit d2595d86e5 for (buggy) Cxt codecs, so better to limit the sleep to the problem hardware. For the second 10ms sleep, the HDA spec says: Power State[1:0]: 00: Node Power state (D0) is fully on. 01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog playback) which must remain fully on. 10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state. 11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software control. Note that any low power state set by software must retain sufficient operational capability to properly respond to subsequent software Power State command. So 10ms is actually the max wait time. It should be safe to remove/reduce it and rely on the loop of 1ms-sleeps. CC: Marc Boucher <marc@linuxant.com> CC: Arjan van de Ven <arjan@linux.intel.com> Signed-off-by: Zhang Rui <rui.zhang@intel.com> Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ASoC: OMAP-McBSP: ASoC interface for McBSP sidetoneIlkka Koskinen
Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones. Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Tony Lindgren <tony@atomide.com>
2010-02-23ASoC: fsi: Modify over/under run error settlementKuninori Morimoto
In current FSI driver, playback function cares only overrun, and capture function cares only underrun. But playback function should had cared about underrun, and capture function should had cared about overrun too. Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23ASoC: OMAP4: Add McPDM platform driverMisael Lopez Cruz
McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Margarita Olaya <x0080101@ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23ASoC: OMAP4: Add support for McPDMCandelaria Villareal, Jorge
McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya <x0080101@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23ASoC: OMAP: data_type and sync_mode configurable in audio dmaMisael Lopez Cruz
Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23ALSA: add support for Macbook Air 2,1 internal speakerReimundo Heluani
Add support for Macbook Air 2,1 (late 2008) internal speaker and headphones. Create a "mba21" model for snd-hda-intel. Signed-off-by: Reimundo Heluani <rheluani@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: usbaudio: consolidate header filesDaniel Mack
Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: usbmixer: bail out early when parsing audio class v2 descriptorsDaniel Mack
This is just a quick hack that needs to be removed once the new units defined by the audio class v2.0 standard are supported. However, it allows using these devices for now, without mixer support. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: usbaudio: implement basic set of class v2.0 parserDaniel Mack
This adds a number of parsers for audio class v2.0. In particular, the following internals are different and now handled by the code: * the number of streaming interfaces is now reported by an interface association descriptor. The old approach using a proprietary descriptor is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the format_type descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than a fixed value. For now, this is worked around by choosing just one of them. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed im the class request command. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: usbaudio: introduce new types for audio class v2Daniel Mack
This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: usbaudio: parse USB descriptors with structsDaniel Mack
In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: hda - enable snoop for Intel Cougar PointSeth Heasley
This patch enables snoop, eliminating static during playback. This patch supersedes the previous Cougar Point audio patch. Signed-off-by: Seth Heasley <seth.heasley@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23ALSA: hda - Remove identical definitions for macmini3 modelTakashi Iwai
The channel mode definitions for macmini3 model are identical with mb5. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22Merge remote branch 'alsa/fixes' into fix/miscTakashi Iwai
2010-02-22ASoC: core: On resume also check the soc device statePeter Ujfalusi
Check the card->codec on soc_resume to detect if the soc device is properly initialized. If the card->codec is NULL, than do not continue the resume operation, since the device is not initialized properly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22ALSA: via82xx: add quirk for D1289 motherboardClemens Ladisch
Add a headphones-only quirk for the Fujitsu Siemens D1289. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de> Cc: <stable@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-22ALSA: usbaudio Mbox support, output onlyChris J Arges
Signed-off-by: Chris J Arges <christopherarges@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22ALSA: hda-intel: Add position_fix quirk for ASUS M2V-MX SE.Paul Menzel
With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1]. Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE. The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker. $ lspci -vvnn | grep -A10 Audio 20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:8290] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel [1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22ALSA: Typo. s/distrubs/disturbs/Paul Menzel
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22ALSA: hda - Clean up Intel Mac unsol codesTakashi Iwai
Use the standard unsol_event callback with each setup callback for IntelMac models with Realtek ALC885 codecs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22ALSA: hda - Add Macmini 3,1 supportLuke Yelavich
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10QDaniel T Chen
BugLink: https://bugs.launchpad.net/bugs/524948 The OR has verified that the existing model=laptop-eapd quirk does not function correctly but instead needs model=3stack. Make this change so that manual corrections to module-init-tools file(s) are not required. Reported-by: Lasse Havelund <lasse@havelund.org> CC: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18ALSA: cs46xx - fix some typosFlorian Zumbiehl
Signed-off-by: Florian Zumbiehl <florz@florz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18ALSA: cs46xx - Do test writes to register AC97_REC_GAIN inFlorian Zumbiehl
snd_cs46xx_codec_reset() bypassing the register cache, so as to not clobber the cached register value during resume. Signed-off-by: Florian Zumbiehl <florz@florz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-17Merge branch 'omap-fixes-for-linus' into omap-for-linusTony Lindgren
2010-02-17ASoC: Make pmdown_time a longMark Brown
Fixes a warning. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-17ASoC: TWL4030: Use codec defaults for Headset initial configurationPeter Ujfalusi
Disable the amplifiers for the headset outputs, and do not select routings by default to the headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-17Merge branch 'fix/misc' into topic/miscTakashi Iwai
Conflicts: sound/pci/hda/patch_realtek.c
2010-02-17Merge remote branch 'alsa/fixes' into fix/miscTakashi Iwai
2010-02-17ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50Giuliano Pochini
This patch fixes a division by zero error in the irq handler. There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-16ASoC: tlv320dac33: Correct the OSCSET calculationPeter Ujfalusi
OSCSET calculation was not correct in case of 44.1KHz sampling rate. With small adjustment both 48 and 44.1 KHz calculation now gives the correct value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-16ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playbackPeter Ujfalusi
In repeated playback the FIFOFLUSH bit remained set, and never has been cleared. Clear it during the setup phase. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-16ASoC: Make pmdown_time runtime configurableMark Brown
Provide a sysfs file allowing userspace to inspect and change the pmdown_time setting at runtime. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-16ASoC: Make pmdown_time a per-card settingMark Brown
Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-16ALSA: usbmixer - use MAX_ID_ELEMS where possibleJaroslav Kysela
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: usbmixer - add usb_id value to usbmixer proc fileJaroslav Kysela
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: pcm core - fix fifo_size channels interval checkJaroslav Kysela
Signed-off-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@kernel.org>
2010-02-16ALSA: usbmixer - introduce /proc/asound/card#/usbmixer fileJaroslav Kysela
The usbmixer proc file contains mapping between ALSA control API and USB mixer control units. The purpose of this file is for debugging and a problem diagnostics. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16Merge branch 'topic/misc' of ↵Jaroslav Kysela
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel
2010-02-16Merge branch 'fixes' into develJaroslav Kysela
2010-02-16ALSA: USB MIDI support for Access Music VirusTISebastien Alaiwan
Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16ALSA: usb-audio: reduce MIDI packet size to work around broken firmwareClemens Ladisch
Extend the list of devices whose firmware does not expect more than one USB MIDI packet in one USB packet. bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-16Merge branch 'fix/hda' of ↵Linus Torvalds
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda - Correct ASUA blacklist for MSI brokenness
2010-02-15omap3: Replace ARCH_OMAP34XX with ARCH_OMAP3Tony Lindgren
Replace ARCH_OMAP34XX with ARCH_OMAP3 Signed-off-by: Tony Lindgren <tony@atomide.com>
2010-02-15omap2: Convert ARCH_OMAP24XX to ARCH_OMAP2Tony Lindgren
Convert ARCH_OMAP24XX to ARCH_OMAP2 Signed-off-by: Tony Lindgren <tony@atomide.com>
2010-02-15ALSA: hda - Correct ASUA blacklist for MSI brokennessTakashi Iwai
The MSI blacklist entry for ASUS mobo added in the commit 8ce28d6abff34886d3797b25324c940471b99164 was based on the alsa-info output wrongly posted. Fix the id to the right one now. Reported-by: Sid Boyce <sboyce@blueyonder.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-15ALSA: Echoaudio - Add suspend support #2Giuliano Pochini
This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). Signed-off-by: Giuliano Pochini <pochini@shiny.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v0.10.2 at 2024-07-11 07:19:07 (GMT)