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* Merge branch 'topic/ymfpci' into for-linusTakashi Iwai2009-09-10
|\ | | | | | | | | * topic/ymfpci: sound: ymfpci: increase timer resolution to 96 kHz
| * sound: ymfpci: increase timer resolution to 96 kHzClemens Ladisch2009-08-10
| | | | | | | | | | | | | | | | Allow the interval timer to be programmed with its full 96 kHz precision. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'topic/usb-audio' into for-linusTakashi Iwai2009-09-10
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/usb-audio: ALSA: usb-audio - Fix types taken in min() sound: usb-audio: do not make URBs longer than sync packet interval sound: usb-audio: add MIDI drain callback sound: usb-audio: use multiple output URBs sound: usb-audio: use multiple input URBs sound: usb-audio: Xonar U1 digital output support
| * | ALSA: usb-audio - Fix types taken in min()Takashi Iwai2009-08-11
| | | | | | | | | | | | | | | | | | | | | | | | Fix the compile warning due to different integer types used in min(): sound/usb/usbaudio.c: In function 'init_substream_urbs': sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: usb-audio: do not make URBs longer than sync packet intervalClemens Ladisch2009-08-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | Using more packets in one URB do avoid interrupts does not make sense when we have a sync pipe whose packets generate interrupts more often. Therefore, limit the URB size to the synchronization packet interval. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: usb-audio: add MIDI drain callbackClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | When draining, instead of waiting for fifty milliseconds, just wait for the currently active URBs to complete. This cuts the usual waiting time down to one USB frame, or zero in the common case when there is no URB. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: usb-audio: use multiple output URBsClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | Some newer USB MIDI interfaces use rather small packet sizes, so to get enough bandwidth, we have to be able to send multiple packets in one USB frame, so we have to use multiple URBs. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: usb-audio: use multiple input URBsClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | Some newer USB MIDI interfaces use rather small packet sizes, so to get enough bandwidth, we have to be able to receive multiple packets in one USB frame, so we have to use multiple URBs. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: usb-audio: Xonar U1 digital output supportClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | Add support for the Asus Xonar U1. This device is mostly class compliant, but the digital output requires a vendor-specific request. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'topic/tlv-minmax' into for-linusTakashi Iwai2009-09-10
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | * topic/tlv-minmax: ALSA: usb-audio - Correct bogus volume dB information ALSA: usb-audio - Use the new TLV_DB_MINMAX type ALSA: Add new TLV types for dBwith min/max
| * | | ALSA: usb-audio - Correct bogus volume dB informationTakashi Iwai2009-06-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some USB devices give bogus dB information and it screws up PA. It's better to detect a broken value and correct it in the driver before exposing the value to the outside. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: usb-audio - Use the new TLV_DB_MINMAX typeTakashi Iwai2009-06-17
| | | | | | | | | | | | | | | | | | | | | | | | Use the new TLV_DB_MINMAX type instead of TLV_DB_SCALE. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: Add new TLV types for dBwith min/maxTakashi Iwai2009-06-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'topic/soundcore-preclaim' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/soundcore-preclaim: sound: make OSS device number claiming optional and schedule its removal sound: request char-major-* module aliases for missing OSS devices chrdev: implement __[un]register_chrdev()
| * | | | sound: make OSS device number claiming optional and schedule its removalTejun Heo2009-08-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If any OSS support is enabled, regardless of built-in or module, sound_core claims full OSS major number (that is, the old 0-255 region) to trap open attempts and request sound modules using custom module aliases. This feature is redundant as chrdev already has such mechanism. This preemptive claiming prevents alternative OSS implementation. The custom module aliases are scheduled to be removed and the previous patch made soundcore emit the standard chrdev aliases too to help transition. This patch schedule the feature for removal in a year and makes it optional so that developers and distros can try new things in the meantime without rebuilding the kernel. The pre-claiming can be turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter soundcore.preclaim_oss. As this allows sound minors to be individually grabbed by other users, this patch updates sound_insert_unit() such that if registering individual device region fails, it tries the next available slot. For details on removal plan, please read the entry added by this patch in feature-removal-schedule.txt . Signed-off-by: Tejun Heo <tj@kernel.org> Cc: Alan Cox <alan@lxorguk.ukuu.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | sound: request char-major-* module aliases for missing OSS devicesTejun Heo2009-08-10
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Till now missing OSS devices emitted sound-slot/service-* module alises instead of the standard char-major-* if a missing device number is opened if soundcore is loaded. The custom module aliases don't have any inherent benefit than backward compatibility. sound-slot/service-* module aliases is scheduled to be removed and to help the transition this patch makes soundcore emit the standard module alises along with the custom ones. Signed-off-by: Tejun Heo <tj@kernel.org> Cc: Alan Cox <alan@lxorguk.ukuu.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'topic/snd-printk' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/snd-printk: ALSA: Fixed a typo of printk() ALSA: Add debug module option ALSA: core - strip too long file names in snd_print*()
| * | | | ALSA: Add debug module optionTakashi Iwai2009-08-27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add debug module option to snd core. This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE is set, you can suppress the debug messages by giving or changing this parameter to a lower value. debug=0 means no debug messsages. As default, it's set to the verbose level 2. Since this option can be changed dynamically via sysfs file, you can suppress the verbose debug messages on the fly, which wasn't possible before. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: core - strip too long file names in snd_print*()Takashi Iwai2009-08-27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When modules are built with M= option, they pass long file paths to __FILE__. This results in ugly outputs of snd_print*() when CONFIG_SND_VERBOSE_PRINTK is set. This patch adds a check of the path and strips the leading path dirs if the file name is an absolute path to improve the readability of logs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'topic/pcm-estrpipe-in-pm' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | * topic/pcm-estrpipe-in-pm: ALSA: pcm - Tell user that stream to be rewound is suspended
| * | | | | ALSA: pcm - Tell user that stream to be rewound is suspendedLubomir Rintel2009-08-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Return STRPIPE instead of EBADF when userspace attempts to rewind of forward a stream that was suspended in meanwhile, so that it can be recovered by snd_pcm_recover(). This was causing Pulseaudio to unload the ALSA sink module under a race condition when it attempted to rewind the stream right after resume from suspend, before writing to the stream which would cause it to revive the stream otherwise. Tested to work with Pulseaudio patched to attempt to snd_pcm_recover() upon receiving an error from snd_pcm_rewind(). Signed-off-by: Lubomir Rintel <lkundrak@v3.sk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | Merge branch 'topic/pcm-drain-nonblock' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/pcm-drain-nonblock: ALSA: pcm - Increase protocol version ALSA: pcm - Fix drain behavior in non-blocking mode
| * | | | | | ALSA: pcm - Fix drain behavior in non-blocking modeTakashi Iwai2009-08-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current PCM core has the following problems regarding PCM draining in non-blocking mode: - the current f_flags isn't checked in snd_pcm_drain(), thus changing the mode dynamically via snd_pcm_nonblock() after open doesn't work. - calling drain in non-blocking mode just return -EAGAIN error, but doesn't provide any way to sync with draining. This patch fixes these issues. - check file->f_flags in snd_pcm_drain() properly - when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state but quits ioctl immediately without waiting the whole drain; the caller can sync the drain manually via poll() Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | Merge branch 'topic/oxygen' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/oxygen: sound: oxygen: work around MCE when changing volume
| * | | | | | | sound: oxygen: work around MCE when changing volumeClemens Ladisch2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Clemens Ladisch <clemens@ladisch.de> Tested-by: Johann Messner <johann.messner at jku.at> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | Merge branch 'topic/oss' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/oss: ALSA: allocation may fail in snd_pcm_oss_change_params() sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma() sound: fix OSS MIDI output data loss
| * | | | | | | | ALSA: allocation may fail in snd_pcm_oss_change_params()Roel Kluin2009-08-31
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allocation may fail, show if it did. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> [Additional fix for invalid runtime->oss.prepare flag set by tiwai] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()Roel Kluin2009-08-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since !LI_CCFG_* evaluates to 0, this did not change anything to cfgval and ctlval. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | sound: fix OSS MIDI output data lossClemens Ladisch2009-08-10
| | |_|_|_|/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the 2.1.6 kernel, the output loop in midi_poll() was changed to enable interrupts during the outputc() call. Unfortunately, the check whether the device has accepted the current byte ("ok") was moved behind the code that removes the byte from the output queue, so one byte would be lost every time the hardware FIFO is full. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | Merge branch 'topic/misc' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/misc: ALSA: Remove unneeded ifdef from sound/core.h ALSA: Remove struct snd_monitor_file from public sound/core.h ALSA: Release v1.0.21
| * | | | | | | | ALSA: Remove struct snd_monitor_file from public sound/core.hTakashi Iwai2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | | | | | | | | Merge branch 'topic/midi' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/midi: sound: rawmidi: disable active-sensing-on-close by default sound: seq_oss_midi: remove magic numbers sound: seq_midi: do not send MIDI reset when closing seq-midi: always log message on output overrun
| * | | | | | | | | sound: rawmidi: disable active-sensing-on-close by defaultClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sending an Active Sensing message when closing a port can interfere with the following data if the port is reopened and a note-on is sent before the device's timeout has elapsed. Therefore, it is better to disable this setting by default. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | sound: seq_oss_midi: remove magic numbersClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of using magic numbers for the controlles sent when resetting a port, use the symbols from asoundef.h. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | sound: seq_midi: do not send MIDI reset when closingClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sending a MIDI reset message when closing a port is wrong because we only want to shut the device up, not to reset all settings. Furthermore, many devices ignore this message. Fortunately, the RawMIDI layer already shuts the device up, so we can ignore this matter here. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | seq-midi: always log message on output overrunClemens Ladisch2009-07-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It turns out that the main cause of output buffer overruns is not slow drivers but applications that generate too many messages. Therefore, it makes more sense to make that error message always visible, and to rate-limit it. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | | | Merge branch 'topic/ice1724-pm' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/ice1724-pm: ALSA: ice1724 - Fix section mismatch ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
| * | | | | | | | | | ALSA: ice1724 - Fix section mismatchTakashi Iwai2009-07-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now snd_vt1724_chip_reset() is used in the resume callback, thus it cannot be __devinit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2Igor Chernyshev2009-06-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I've built a small HTPC and had to add suspend/resume support in ice1724 driver. There seem to be 3 existing bugs related to that: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314 Due to hardware (un)availability, I only enabled the fix for Audiotrak Prodigy HD2 card, which is installed in my HTPC. However, most of my code should be reusable in the future on other ice1724-based cards as well (as long as people add card-specific peices of code). The fix is currently based on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel). Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | | | | Merge branch 'topic/hdsp' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/hdsp: ALSA: hdsp - allow proc reporting with disconnected io box
| * | | | | | | | | | | ALSA: hdsp - allow proc reporting with disconnected io boxTim Blechmann2009-08-12
| | |_|_|/ / / / / / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | the hdsp driver refuses to report any information via the proc interface, if the io box is not connected. with this patch, the content of the control and status registers is printed before the iobox check. Signed-off-by: Tim Blechmann <tim@klingt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | | | | Merge branch 'topic/hda' into for-linusTakashi Iwai2009-09-10
|\ \ \ \ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * topic/hda: (92 commits) ALSA: hda - Use auto model for HP laptops with ALC268 codec ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital ALSA: hda - Add support of Alienware M17x laptop ALSA: hda - Remove dead codes from patch_sigmatel.c ALSA: hda - Fix input source selection of IDT92HD73xx ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT ALSA: hda - Unmute docking line-out as default with AD1984A codec ALSA: hda - Add another entry for Nvidia HDMI device ALSA: hda - Add missing GPIO initialization for AD1984A laptop model ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model ALSA: hda - Fix ALC268/ALC269 headphone pin routing ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs ALSA: hda - Add more quirk for HP laptops with AD1984A ALSA: hda - Add / fix model entries for HD-audio driver ALSA: hda - Add full audio support on Acer Aspire 7730G notebook ALSA: hda - Improve auto-cfg mixer name for ALC662 ALSA: hda - Improve auto-cfg mixer name for ALC861-VD ALSA: hda - Improve auto-cfg mixer name for ALC262 ALSA: hda - Improve auto-cfg mixer name for ALC260 ALSA: hda - Improve auto-cfg mixer name for ALC880 ...
| * | | | | | | | | | | ALSA: hda - Use auto model for HP laptops with ALC268 codecTakashi Iwai2009-09-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Jens Jorgensen <jbj1@ultraemail.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digitalddiaz@cenditel.gob.ve2009-09-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Dhionel Diaz <ddiaz@cenditel.gob.ve> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Add support of Alienware M17x laptopTakashi Iwai2009-09-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Remove dead codes from patch_sigmatel.cTakashi Iwai2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Fix input source selection of IDT92HD73xxTakashi Iwai2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECTTakashi Iwai2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the old dead CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Unmute docking line-out as default with AD1984A codecTakashi Iwai2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | | | | ALSA: hda - Add another entry for Nvidia HDMI deviceTakashi Iwai2009-09-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Added another entry for Nvidia HDMI device (10de:0003). Reference: kernel bug#14097 http://bugzilla.kernel.org/show_bug.cgi?id=14097 Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v1.2.2 (git 2.25.0) at 2024-09-20 02:31:06 -0400