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* ALSA: hda - Disable the sanity check in snd_hda_add_pincfg()Takashi Iwai2013-04-181-0/+6
| | | | | | | | | | | | When pin default configs are overridden via patch option, these are evaluated before fixups are applied. Since some fixups change the whole codec trees and/or add pins dynamically, this sanity check might not pass when pins aren't present at the time the function is called. We may reorder the execution, but an easier fix is simply to disable this sanity check. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - fix error return code in patch_alc662()Wei Yongjun2013-04-181-1/+2
| | | | | | | | Fix to return a negative error code from the error handling case instead of 0, as returned elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Don't call vmaster hook when bus->shutdown is setTakashi Iwai2013-04-171-0/+5
| | | | | | | | | | | | | | | | The flag bus->shutdown implies that the control elements might have been already destroyed. When a codec is resumed at this state and tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would refer to a non-existing object, resulting in Oops in the end. This patch just adds a check of the flag in the caller side for avoiding such a crash. Though, the best would be to clear hook->sw_kctl by the destructor of the corresponding ctl element, but vmaster uses its own private_free, it can't be done easily. So let it be for a while. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - fixup D3 pin and right channel mute on Haswell HDMI audioDavid Henningsson2013-04-171-0/+38
| | | | | | | | | | | | | | When graphics initializes the HDMI chip, sometimes this leads to pins going into D3 and right channel being muted. If the audio driver finishes initialization before the graphic driver does, this situation becomes permanent. This is a workaround that checks for this situation and corrects it on playback prepare. It has been verified working on at least one machine. BugLink: https://bugs.launchpad.net/bugs/1167270 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Use the primary DAC for all aamix outputsTakashi Iwai2013-04-161-4/+7
| | | | | | | | | | When setting up the aamix output paths, use the primary DAC instead of the individual DAC for each output as default. Otherwise multiple DACs will be turned on for a single aamix widget, which results in doubly or more volumes, because the duplicated signals will be sent through all these DACs for a single stream. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix aamix activation with loopback control on VIA codecsTakashi Iwai2013-04-161-6/+16
| | | | | | | | | | | | | | | | | | | | | When we have a loopback mixer control, this should manage the state whether the output paths include the aamix or not. But the current code blindly initializes the output paths with aamix = true, thus the aamix is enabled unless the loopback mixer control is changed. Also, update_aamix_paths() called by the loopback mixer control put callback invokes snd_hda_activate_path() with aamix = true even for disabling the mixing. This leaves the aamix path even though the loopback control is turned off. This patch fixes these issues: - Introduced aamix_default() helper to indicate whether with_aamix is true or false as default - Fix the argument in update_aamix_paths() for disabling loopback Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Cc: <stable@vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add codec delay to the capture time stamp.Dylan Reid2013-04-161-2/+5
| | | | | | | | | For capture, the delay through the codec contributes to the time stamp of the sample recorded at the A to D. Rename the codec time stamp function appropriately. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-v3.10' of ↵Takashi Iwai2013-04-154-7/+7
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.10 A bunch of changes here, the most interesting one subsystem wise being Morimoto-san's work to create snd_soc_component which doesn't do much for now but will be pretty important going forwards: - Add a new component object type which will form the basis of moving to a more generic handling of SoC and off-SoC components, contributed by Kuninori Morimoto. - A fairly large set of cleanups for the dmaengine integration from Lars-Peter Clausen, starting to move towards being able to have a generic driver based on the library. - Performance optimisations to DAPM from Ryo Tsutsui. - Support for mixer control sharing in DAPM from Stephen Warren. - Multiplatform ARM cleanups from Arnd Bergmann. - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
| * ALSA: hda/generic - fix uninitialized variableJiri Slaby2013-04-051-1/+1
| | | | | | | | | | | | | | | | | | changed is not initialized in path_power_down_sync, but it is expected to be false in case no change happened in the loop. So set it to false. Signed-off-by: Jiri Slaby <jslaby@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Revert "ALSA: hda - Allow power_save_controller option override DCAPS"Takashi Iwai2013-04-041-4/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f. The commit [6ab317419c: ALSA: hda - Allow power_save_controller option override DCAPS] changed the behavior of power_save_controller so that it can override the driver capability. This assumed that this option is rarely changed dynamically unlike power_save option. Too naive. It turned out that the user-space power-management tool tries to set power_save_controller option to 1 together with power_save option without knowing what's actually doing. This enabled forcibly the runtime PM of the controller, which is known to be broken om many chips thus disabled as default. So, the only sane fix is to revert this commit again. It was intended to ease debugging/testing for runtime PM enablement, but obviously we need another way for it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171 Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - fix typo in proc outputDavid Henningsson2013-04-041-1/+1
| | | | | | | | | | | | | | | | Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Enabling Realtek ALC 671 codecRainer Koenig2013-04-041-1/+3
| | | | | | | | | | | | | | | | | | * Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix headset mic support for Asus X101CHDavid Henningsson2013-04-151-0/+56
| | | | | | | | | | | | | | | | | | | | | | | | With this patch, a TRRS headset mic cannot be successfully detected on the Asus X101CH, and we can also distinguish between headphone and headset automatically. Buglink: https://bugs.launchpad.net/bugs/1169138 Co-authored-by: Kailang <kailang@realtek.com> Tested-by: Luis Henriques <luis.henriques@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Implement headset jack functionality for some Dell hwDavid Henningsson2013-04-151-0/+386
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On some machines, there is a headset jack that can support both headphone, headsets (of both CTIA and OMTP type) and mic-in. On other machines, the headset jack supports headphone, headsets (both CTIA and OMTP), but not mic-in. This patch implements that functionality as different capture sources. Buglink: https://bugs.launchpad.net/bugs/1169143 Tested-by: David Chen <david.chen@canonical.com> Co-authored-by: Kailang <kailang@realtek.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Handle Headphone Mic jack more genericDavid Henningsson2013-04-112-24/+30
| | | | | | | | | | | | | | | | Now that we have a flag for headphone mics, we can use that flag in the jack creation instead of creating the jack manually. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - add some small convenience functions to auto parserDavid Henningsson2013-04-111-0/+21
| | | | | | | | | | | | | | | | | | | | I never liked that we move our speaker and hp pins to line out if there are not any line outs; but now that we do, add some convenience functions to find hp and speaker pins even if they have been moved. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - allow "Headphone Mic" parser flagDavid Henningsson2013-04-113-15/+62
| | | | | | | | | | | | | | | | | | | | | | | | This allows a specific mic to get the "Headphone Mic" name, in addition to the existing "Headset Mic" name. Also, it allows for a special mark: if the sequence number is set to 0xc, that's an indication to prefer it for headset mic, and if it's set to 0xd, that's an indication to prefer it for headphone mic. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Apply codec delay to wallclock.Dylan Reid2013-04-091-0/+18
| | | | | | | | | | | | | | | | | | For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/realtek - Add a quirk for AC700 Chromebook.Dylan Reid2013-04-071-1/+17
| | | | | | | | | | | | | | | | | | | | Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.Dylan Reid2013-04-051-1/+18
| | | | | | | | | | | | | | The Stumpy ChromeBox needs its pin configs fixed up. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/ca0132 - Update latency based on DSP state.Dylan Reid2013-04-051-2/+53
| | | | | | | | | | | | | | | | | | | | | | The DSP in the CA0132 codec adds a variable latency to audio depending on what processing is being done. Add a new patch op to return that latency for capture and playback streams. The latency is determined by which blocks are enabled and knowing how much latency is added by each block. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Introduce get_delay codec PCM opsTakashi Iwai2013-04-052-4/+17
| | | | | | | | | | | | | | | | | | | | Add a new codec PCM ops, get_delay(), to obtain the codec/stream- specific PCM delay count. When it's NULL, nothing changes. This new feature was requested for CA0132, which has significant delays in the path depending on the running DSP code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-04-032-2/+2
|\| | | | | | | | | Back-merge for cleaning up usb-audio code the recent commit modified, and further UAC2 autoclock patches.
| * ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin2013-04-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin2013-04-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: stable@vger.kernel.org Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/ca0132 - Delay HP amp turnon.Chih-Chung Chang2013-04-021-4/+23
| | | | | | | | | | | | | | | | | | | | | | Turing on the headphone amp interferes with the impedance measurement used to detect a TRRS style headset microphone. Delay the HP turn on until 500ms after the jack is detected, allowing the mic detection state machine to run to completion. Signed-off-by: Chih-Chung Chang <chihchung@chromium.org> Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - VIA prefers side surrounds over HPTakashi Iwai2013-03-221-0/+20
| | | | | | | | | | | | | | | | | | | | | | | | The recent fix for the independent HP reduced the availability of the side surround output, because there are only 4 DACs for 7.1 and a HP outputs. Adjust the badness tables for VIA so that 7.1 outputs are activated for the cost of missing independent HP. Once when we implement the dynamic DAC switching to multiple outputs, this conflicts will be eased in future... Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Lower the badness for independent HP penaltyTakashi Iwai2013-03-221-1/+1
| | | | | | | | | | | | | | The lack of independent HP mode shouldn't be too bad, but currently its badness is set a bit too high. Let's lower it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Allow codec drivers to give own badness tablesTakashi Iwai2013-03-222-14/+29
| | | | | | | | | | | | | | | | | | | | | | The standard badness values don't seem to fit to all preferences. Some configuration prefer the side output over the headphone, some want the speaker over the surround, etc. This patch moves the badness table pointers into hda_gen_spec, so that the codec driver can override them. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-03-224-26/+158
|\| | | | | | | | | | | | | | | | | | | | | | | | | Merge back for-linus branch for the badness table adjustment for VIA codecs * for-linus: ALSA: hda - Fix DAC assignment for independent HP ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader ALSA: hda - Fix typo in checking IEC958 emphasis bit ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() ALSA: snd-usb: mixer: propagate errors up the call chain ALSA: usb: Parse UAC2 extension unit like for UAC1 ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
| * ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai2013-03-211-0/+46
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loaderTakashi Iwai2013-03-201-23/+109
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix typo in checking IEC958 emphasis bitTakashi Iwai2013-03-201-1/+1
| | | | | | | | | | | | | | | | | | | | There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart <martin.weishart@telosalliance.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driverTakashi Iwai2013-03-181-2/+2
| | | | | | | | | | | | | | | | I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Enable "Headset Mic" name for some Dell Latitude devicesDavid Henningsson2013-03-211-1/+5
| | | | | | | | | | | | | | | | Now that we have a "Headset Mic" name, let's use it for some devices we know for sure has a headset mic jack. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Introduce "Headset Mic" nameDavid Henningsson2013-03-212-2/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Add GPIO-based LED support on HP desktop machinesTakashi Iwai2013-03-181-0/+63
| | | | | | | | | | | | | | | | The new HP desktop machines have Realtek codecs and their LEDs are controlled via GPIO as for many laptop models. Add similar hooks as well as in patch_sigmatel.c for controlling LEDs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Make the resume of digital beep setup properTakashi Iwai2013-03-181-1/+1
| | | | | | | | | | | | | | | | The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2 should be executed at resume as well. Use the cached write for it being performed automatically at resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix power-saving during playing beep soundTakashi Iwai2013-03-182-11/+27
| | | | | | | | | | | | | | | | | | | | | | While playing the digital beep tone, the codec shouldn't be turned off. This patch adds proper snd_hda_power_up()/down() calls at each time when the beep is played or off. Also, this fixes automatically an unnecessary codec power-up at detaching the beep device when the beep isn't being played. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Move beep attach/detach calls in hda_generic.cTakashi Iwai2013-03-186-128/+72
| | | | | | | | | | | | | | | | | | Instead of calling snd_hda_attach_beep_device() and snd_hda_detach_beep_device() in each codec driver, move them to the generic parser. The codec driver just needs to set spec->beep_nid for activating the digital beep. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-03-188-33/+94
|\| | | | | | | Back-merged for refactoring beep stuff.
| * ALSA: hda/cirrus - Fix the digital beep registrationTakashi Iwai2013-03-181-4/+4
| | | | | | | | | | | | | | | | The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix missing beep detach in patch_conexant.cTakashi Iwai2013-03-181-1/+7
| | | | | | | | | | | | | | | | This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix missing EAPD/GPIO setup for Cirrus codecsTakashi Iwai2013-03-151-0/+4
| | | | | | | | | | | | | | | | During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Remove extra setting of dsp_state.Dylan Reid2013-03-151-9/+6
| | | | | | | | | | | | | | | | spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Check download state of DSP.Dylan Reid2013-03-151-2/+3
| | | | | | | | | | | | | | | | | | | | | | Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda/ca0132 - Check if dspload_image succeeded.Dylan Reid2013-03-151-2/+6
| | | | | | | | | | | | | | | | | | | | If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Disable IDT eapd_switch if there are no internal speakersDavid Henningsson2013-03-141-0/+29
| | | | | | | | | | | | | | | | | | If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix snd_hda_get_num_raw_conns() to return a correct valueTakashi Iwai2013-03-121-10/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the connection list expansion in hda_codec.c and hda_proc.c, the value returned from snd_hda_get_num_raw_conns() is used as the array size to store the connection list. However, the function returns simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the widget list with ranges isn't considered there. Thus it may return a smaller size than the actual list, which results in -ENOSPC in snd_hda_get_raw_conections(). This patch fixes the bug by parsing the connection list correctly also for snd_hda_get_num_raw_conns(). Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: asihpi - fix potential NULL pointer dereferenceWei Yongjun2013-03-121-1/+2
| | | | | | | | | | | | | | The dereference should be moved below the NULL test. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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