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* ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chipsTakashi Iwai2019-09-161-4/+2
| | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 2756d9143aa517b97961e85412882b8ce31371a6 ] It turned out that the recent Intel HD-audio controller chips show a significant stall during the system PM resume intermittently. It doesn't happen so often and usually it may read back successfully after one or more seconds, but in some rare worst cases the driver went into fallback mode. After trial-and-error, we found out that the communication stall seems covered by issuing the sync after each verb write, as already done for AMD and other chipsets. So this patch enables the write-sync flag for the recent Intel chips, Skylake and onward, as a workaround. Also, since Broxton and co have the very same driver flags as Skylake, refer to the Skylake driver flags instead of defining the same contents again for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901 Reported-and-tested-by: Todd Brandt <todd.e.brandt@linux.intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda - Don't resume forcibly i915 HDMI/DP codecTakashi Iwai2019-09-163-3/+13
| | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 4914da2fb0c89205790503f20dfdde854f3afdd8 ] We apply the codec resume forcibly at system resume callback for updating and syncing the jack detection state that may have changed during sleeping. This is, however, superfluous for the codec like Intel HDMI/DP, where the jack detection is managed via the audio component notification; i.e. the jack state change shall be reported sooner or later from the graphics side at mode change. This patch changes the codec resume callback to avoid the forcible resume conditionally with a new flag, codec->relaxed_resume, for reducing the resume time. The flag is set in the codec probe. Although this doesn't fix the entire bug mentioned in the bugzilla entry below, it's still a good optimization and some improvements are seen. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201901 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentreHui Wang2019-09-161-0/+2
| | | | | | | | | | | | | | | | | | | | | | commit 2a36c16efab254dd6017efeb35ad88ecc96f2328 upstream. This ThinkCentre machine has a new realtek codec alc222, it is not in the support list, we add it in the realtek.c then this machine can apply FIXUPs for the realtek codec. And this machine has two front mics which can't be handled by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't work on this machine. Fortunately another FIXUP ALC283_FIXUP_HEADSET_MIC also can change the location for one of the two mics on this machine. Link: https://lore.kernel.org/r/20190904055327.9883-1-hui.wang@canonical.com Signed-off-by: Hui Wang <hui.wang@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FLJian-Hong Pan2019-09-161-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 60083f9e94b2f28047d71ed778adf89357c1a8fb upstream. Original pin node values of ASUS UX431FL with ALC294: 0x12 0xb7a60140 0x13 0x40000000 0x14 0x90170110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x90170111 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4066852d 0x1e 0x411111f0 0x1f 0x411111f0 0x21 0x04211020 1. Has duplicated internal speakers (0x14 & 0x17) which makes the output route become confused. So, the output volume cannot be changed by setting. 2. Misses the headset mic pin node. This patch disables the confusing speaker (NID 0x14) and enables the headset mic (NID 0x19). Link: https://lore.kernel.org/r/20190902100054.6941-1-jian-hong@endlessm.com Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Add quirk for HP Pavilion 15Sam Bazley2019-09-161-0/+1
| | | | | | | | | | | | | | | commit d33cd42d86671bed870827aa399aeb9f1da74119 upstream. HP Pavilion 15 (AMD Ryzen-based model) with 103c:84e7 needs the same quirk like HP Envy/Spectre x360 for enabling the mute LED over Mic3 pin. [ rearranged in the SSID number order by tiwai ] Signed-off-by: Sam Bazley <sambazley@fastmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Fix overridden device-specific initializationTakashi Iwai2019-09-163-1/+5
| | | | | | | | | | | | | | | | | | | | | | | commit 89781d0806c2c4f29072d3f00cb2dd4274aabc3d upstream. The recent change to shuffle the codec initialization procedure for Realtek via commit 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") caused the silent output on some machines. This change was supposed to be safe, but it isn't actually; some devices have quirk setups to override the EAPD via COEF or BTL in the additional verb table, which is applied at the beginning of snd_hda_gen_init(). And this EAPD setup is again overridden in alc_auto_init_amp(). For recovering from the regression, tell snd_hda_gen_init() not to apply the verbs there by a new flag, then apply the verbs in alc_init(). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204727 Fixes: 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fix potential endless loop at applying quirksTakashi Iwai2019-09-161-2/+2
| | | | | | | | | | | | | | | commit 333f31436d3db19f4286f8862a00ea1d8d8420a1 upstream. Since the chained quirks via chained_before flag is applied before the depth check, it may lead to the endless recursive calls, when the chain were set up incorrectly. Fix it by moving the depth check at the beginning of the loop. Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* Revert "ASoC: Fail card instantiation if DAI format setup fails"Greg Kroah-Hartman2019-09-061-5/+2
| | | | | | | | | | | | | | | | | This reverts commit 714a8438fc8ae88aa22c25065e241bce0260db13 which is commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b upstream. Mark Brown writes: I nacked this patch when Sasha posted it - it only improves diagnostics and might make systems that worked by accident break since it turns things into a hard failure, it won't make anything that didn't work previously work. Reported-by: Mark Brown <broonie@kernel.org> Cc: Ricard Wanderlof <ricardw@axis.com> Cc: Sasha Levin <sashal@kernel.org> Link: https://lore.kernel.org/lkml/20190904181027.GG4348@sirena.co.uk Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604Takashi Iwai2019-09-061-0/+1
| | | | | | | | | | | | | commit 1a15718b41df026cffd0e42cfdc38a1384ce19f9 upstream. Behringer UFX1604 requires the similar quirk to apply implicit fb like another Behringer model UFX1204 in order to fix the noisy playback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()Takashi Iwai2019-09-061-4/+4
| | | | | | | | | | | | | | | | | | | | | commit 6de3c9e3f6b3eaf66859e1379b3f35dda781416b upstream. The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: seq: Fix potential concurrent access to the deleted poolTakashi Iwai2019-09-063-2/+20
| | | | | | | | | | | | | | | commit 75545304eba6a3d282f923b96a466dc25a81e359 upstream. The input pool of a client might be deleted via the resize ioctl, the the access to it should be covered by the proper locks. Currently the only missing place is the call in snd_seq_ioctl_get_client_pool(), and this patch papers over it. Reported-by: syzbot+4a75454b9ca2777f35c7@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fixes inverted Conexant GPIO mic mute ledJeronimo Borque2019-09-061-8/+9
| | | | | | | | | | | | | | | | | | | | commit f9ef724d4896763479f3921afd1ee61552fc9836 upstream. "enabled" parameter historically referred to the device input or output, not to the led indicator. After the changes added with the led helper functions the mic mute led logic refers to the led and not to the mic input which caused led indicator to be negated. Fixing logic in cxt_update_gpio_led and updated cxt_fixup_gpio_mute_hook Also updated debug messages to ease further debugging if necessary. Fixes: 184e302b46c9 ("ALSA: hda/conexant - Use the mic-mute LED helper") Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jeronimo Borque <jeronimo@borque.com.ar> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: line6: Fix memory leak at line6_init_pcm() error pathTakashi Iwai2019-09-061-9/+9
| | | | | | | | | | | | | | | commit 1bc8d18c75fef3b478dbdfef722aae09e2a9fde7 upstream. I forgot to release the allocated object at the early error path in line6_init_pcm(). For addressing it, slightly shuffle the code so that the PCM destructor (pcm->private_free) is assigned properly before all error paths. Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Check mixer unit bitmap yet more strictlyTakashi Iwai2019-09-061-8/+28
| | | | | | | | | | | | | | | | | | | | | | commit f9f0e9ed350e15d51ad07364b4cf910de50c472a upstream. The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a variable size depending on both input and output pins. Its size is to fit with input * output bits. The problem is that the input size can't be determined simply from the unit descriptor itself but it needs to parse the whole connected sources. Although the uac_mixer_unit_get_channels() tries to check some possible overflow of this bitmap, it's incomplete due to the lack of the evaluation of input pins. For covering possible overflows, this patch adds the bitmap overflow check in the loop of input pins in parse_audio_mixer_unit(). Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: ti: davinci-mcasp: Correct slot_width posed constraintPeter Ujfalusi2019-08-291-9/+34
| | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 1e112c35e3c96db7c8ca6ddaa96574f00c06e7db ] The slot_width is a property for the bus while the constraint for SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format. Applying slot_width constraint to sample_bits works most of the time, but it will blacklist valid formats in some cases. With slot_width 24 we can support S24_3LE and S24_LE formats as they both look the same on the bus, but a a 24 constraint on sample_bits would not allow S24_LE as it is stored in 32bits in memory. Implement a simple hw_rule function to allow all formats which require less or equal number of bits on the bus as slot_width (if configured). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: rockchip: Fix mono captureCheng-Yi Chiang2019-08-291-3/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 789e162a6255325325bd321ab0cd51dc7e285054 ] This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0. Revert "ASoC: rockchip: i2s: Support mono capture" Previous discussion in https://patchwork.kernel.org/patch/10147153/ explains the issue of the patch. While device is configured as 1-ch, hardware is still generating a 2-ch stream. When user space reads the data and assumes it is a 1-ch stream, the rate will be slower by 2x. Revert the change so 1-ch is not supported. User space can selectively take one channel data out of two channel if 1-ch is preferred. Currently, both channels record identical data. Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org> Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: Fail card instantiation if DAI format setup failsRicard Wanderlof2019-08-291-2/+5
| | | | | | | | | | | | | [ Upstream commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b ] If the DAI format setup fails, there is no valid communication format between CPU and CODEC, so fail card instantiation, rather than continue with a card that will most likely not function properly. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walksCharles Keepax2019-08-291-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 8dd26dff00c0636b1d8621acaeef3f6f3a39dd77 ] DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda - Let all conexant codec enter D3 when rebootingHui Wang2019-08-251-9/+0
| | | | | | | | | | | | | | | | | | | | | commit 401714d9534aad8c24196b32600da683116bbe09 upstream. We have 3 new lenovo laptops which have conexant codec 0x14f11f86, these 3 laptops also have the noise issue when rebooting, after letting the codec enter D3 before rebooting or poweroff, the noise disappers. Instead of adding a new ID again in the reboot_notify(), let us make this function apply to all conexant codec. In theory make codec enter D3 before rebooting or poweroff is harmless, and I tested this change on a couple of other Lenovo laptops which have different conexant codecs, there is no side effect so far. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Add a generic reboot_notifyHui Wang2019-08-254-15/+22
| | | | | | | | | | | | | | | | commit 871b9066027702e6e6589da0e1edd3b7dede7205 upstream. Make codec enter D3 before rebooting or poweroff can fix the noise issue on some laptops. And in theory it is harmless for all codecs to enter D3 before rebooting or poweroff, let us add a generic reboot_notify, then realtek and conexant drivers can call this function. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fix a memory leak bugWenwen Wang2019-08-251-1/+1
| | | | | | | | | | | | | | | | | commit cfef67f016e4c00a2f423256fc678a6967a9fc09 upstream. In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc(). Then, the pin widgets in 'codec' are parsed. However, if the parsing process fails, 'spec' is not deallocated, leading to a memory leak. To fix the above issue, free 'spec' before returning the error. Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Apply workaround for another AMD chip 1022:1487Takashi Iwai2019-08-251-0/+3
| | | | | | | | | | | | | | commit de768ce45466f3009809719eb7b1f6f5277d9373 upstream. MSI MPG X570 board is with another AMD HD-audio controller (PCI ID 1022:1487) and it requires the same workaround applied for X370, etc (PCI ID 1022:1457). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unitHui Peng2019-08-251-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit daac07156b330b18eb5071aec4b3ddca1c377f2c upstream. The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. Reported-by: Hui Peng <benquike@gmail.com> Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Cc: <stable@vger.kernel.org> Signed-off-by: Hui Peng <benquike@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_termHui Peng2019-08-251-8/+27
| | | | | | | | | | | | | | | | | | | | | | | commit 19bce474c45be69a284ecee660aa12d8f1e88f18 upstream. `check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). Reported-by: Hui Peng <benquike@gmail.com> Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Signed-off-by: Hui Peng <benquike@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Add quirk for HP Envy x360Takashi Iwai2019-08-251-0/+1
| | | | | | | | | | | | | commit 190d03814eb3b49d4f87ff38fef26d36f3568a60 upstream. HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)Takashi Iwai2019-08-163-2/+70
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db upstream. A long-time problem on the recent AMD chip (X370, X470, B450, etc with PCI ID 1022:1457) with Realtek codecs is the crackled or distorted sound for capture streams, as well as occasional playback hiccups. After lengthy debugging sessions, the workarounds we've found are like the following: - Set up the proper driver caps for this controller, similar as the other AMD controller. - Correct the DMA position reporting with the fixed FIFO size, which is similar like as workaround used for VIA chip set. - Even after the position correction, PulseAudio still shows mysterious stalls of playback streams when a capture is triggered in timer-scheduled mode. Since we have no clear way to eliminate the stall, pass the BATCH PCM flag for PA to suppress the tsched mode as a temporary workaround. This patch implements the workarounds. For the driver caps, it defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO- corrected position reporting (corresponding to the new position_fix=6) and enforces the SNDRV_PCM_INFO_BATCH flag. Note that the current implementation is merely a workaround. Hopefully we'll find a better alternative in future, especially about removing the BATCH flag hack again. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Don't override global PCM hw info flagTakashi Iwai2019-08-161-4/+2
| | | | | | | | | | | | | | | | | | commit c1c6c877b0c79fd7e05c931435aa42211eaeebaf upstream. The commit bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") introduced the conditional PCM hw info setup, but it overwrites the global azx_pcm_hw object. This will cause a problem if any other HD-audio controller, as it'll inherit the same bit flag although another controller doesn't support that feature. Fix the bug by setting the PCM hw info flag locally. Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hiface: fix multiple memory leak bugsWenwen Wang2019-08-161-3/+8
| | | | | | | | | | | | | | | | | | | | commit 3d92aa45fbfd7319e3a19f4ec59fd32b3862b723 upstream. In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and 'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs. Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails. To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'. Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: firewire: fix a memory leak bugWenwen Wang2019-08-161-1/+1
| | | | | | | | | | | | | | | | | | | | commit 1be3c1fae6c1e1f5bb982b255d2034034454527a upstream. In iso_packets_buffer_init(), 'b->packets' is allocated through kmalloc_array(). Then, the aligned packet size is checked. If it is larger than PAGE_SIZE, -EINVAL will be returned to indicate the error. However, the allocated 'b->packets' is not deallocated on this path, leading to a memory leak. To fix the above issue, free 'b->packets' before returning the error code. Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # v2.6.39+ Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: fix a memory leak bugWenwen Wang2019-08-161-0/+1
| | | | | | | | | | | | | | | | | | commit a67060201b746a308b1674f66bf289c9faef6d09 upstream. In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is allocated through kzalloc() before the execution goto 'found_clock'. However, this structure is not deallocated if the memory allocation for 'pd' fails, leading to a memory leak bug. To fix the above issue, free 'fp->chmap' before returning NULL. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: compress: Be more restrictive about when a drain is allowedCharles Keepax2019-08-161-0/+6
| | | | | | | | | | | | | | [ Upstream commit 3b8179944cb0dd53e5223996966746cdc8a60657 ] Draining makes little sense in the situation of hardware overrun, as the hardware will have consumed all its available samples. Additionally, draining whilst the stream is paused would presumably get stuck as no data is being consumed on the DSP side. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: compress: Don't allow paritial drain operations on capture streamsCharles Keepax2019-08-161-0/+8
| | | | | | | | | | | | | [ Upstream commit a70ab8a8645083f3700814e757f2940a88b7ef88 ] Partial drain and next track are intended for gapless playback and don't really have an obvious interpretation for a capture stream, so makes sense to not allow those operations on capture streams. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: compress: Prevent bypasses of set_paramsCharles Keepax2019-08-161-6/+24
| | | | | | | | | | | | | | | | [ Upstream commit 26c3f1542f5064310ad26794c09321780d00c57d ] Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which allow a transition to SNDRV_PCM_STATE_SETUP. The stream should only be able to move to the setup state once it has received a SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing those ioctls whilst in the open state. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: compress: Fix regression on compressed capture streamsCharles Keepax2019-08-161-5/+11
| | | | | | | | | | | | | | | | | | | | | [ Upstream commit 4475f8c4ab7b248991a60d9c02808dbb813d6be8 ] A previous fix to the stop handling on compressed capture streams causes some knock on issues. The previous fix updated snd_compr_drain_notify to set the state back to PREPARED for capture streams. This causes some issues however as the handling for snd_compr_poll differs between the two states and some user-space applications were relying on the poll failing after the stream had been stopped. To correct this regression whilst still fixing the original problem the patch was addressing, update the capture handling to skip the PREPARED state rather than skipping the SETUP state as it has done until now. Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* sound: fix a memory leak bugWenwen Wang2019-08-161-1/+2
| | | | | | | | | | | | | | | | | | | commit c7cd7c748a3250ca33509f9235efab9c803aca09 upstream. In sound_insert_unit(), the controlling structure 's' is allocated through kmalloc(). Then it is added to the sound driver list by invoking __sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is removed from the list through __sound_remove_unit(). If 'index' is not less than 0, -EBUSY is returned to indicate the error. However, 's' is not deallocated on this execution path, leading to a memory leak bug. To fix the above issue, free 's' before -EBUSY is returned. Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda: Fix 1-minute detection delay when i915 module is not availableSamuel Thibault2019-08-061-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | commit 74bf71ed792ab0f64631cc65ccdb54c356c36d45 upstream. Distribution installation images such as Debian include different sets of modules which can be downloaded dynamically. Such images may notably include the hda sound modules but not the i915 DRM module, even if the latter was enabled at build time, as reported on https://bugs.debian.org/931507 In such a case hdac_i915 would be linked in and try to load the i915 module, fail since it is not there, but still wait for a whole minute before giving up binding with it. This fixes such as case by only waiting for the binding if the module was properly loaded (or module support is disabled, in which case i915 is already compiled-in anyway). Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Signed-off-by: Samuel Thibault <samuel.thibault@ens-lyon.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Add a conexant codec entry to let mute led workHui Wang2019-07-311-0/+1
| | | | | | | | | | | | | | | | commit 3f8809499bf02ef7874254c5e23fc764a47a21a0 upstream. This conexant codec isn't in the supported codec list yet, the hda generic driver can drive this codec well, but on a Lenovo machine with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI to make the leds work. After adding this codec to the list, the driver patch_conexant.c will apply THINKPAD_ACPI to this machine. Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: line6: Fix wrong altsetting for LINE6_PODHD500_1Kai-Heng Feng2019-07-311-1/+1
| | | | | | | | | | | | | | | | | commit 70256b42caaf3e13c2932c2be7903a73fbe8bb8b upstream. Commit 7b9584fa1c0b ("staging: line6: Move altsetting to properties") set a wrong altsetting for LINE6_PODHD500_1 during refactoring. Set the correct altsetting number to fix the issue. BugLink: https://bugs.launchpad.net/bugs/1790595 Fixes: 7b9584fa1c0b ("staging: line6: Move altsetting to properties") Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: ac97: Fix double free of ac97_codec_deviceDing Xiang2019-07-311-9/+4
| | | | | | | | | | | | | | | commit 607975b30db41aad6edc846ed567191aa6b7d893 upstream. put_device will call ac97_codec_release to free ac97_codec_device and other resources, so remove the kfree and other redundant code. Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Signed-off-by: Ding Xiang <dingxiang@cmss.chinamobile.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: apply ALC891 headset fixup to one Dell machineHui Wang2019-07-261-0/+5
| | | | | | | | | | | | commit 4b4e0e32e4b09274dbc9d173016c1a026f44608c upstream. Without this patch, the headset-mic and headphone-mic don't work. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Fixed Headphone Mic can't record on Dell platformKailang Yang2019-07-261-1/+4
| | | | | | | | | | | | | commit fbc571290d9f7bfe089c50f4ac4028dd98ebfe98 upstream. It assigned to wrong model. So, The headphone Mic can't work. Fixes: 3f640970a414 ("ALSA: hda - Fix headset mic detection problem for several Dell laptops") Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: seq: Break too long mutex context in the write loopTakashi Iwai2019-07-261-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | commit ede34f397ddb063b145b9e7d79c6026f819ded13 upstream. The fix for the racy writes and ioctls to sequencer widened the application of client->ioctl_mutex to the whole write loop. Although it does unlock/relock for the lengthy operation like the event dup, the loop keeps the ioctl_mutex for the whole time in other situations. This may take quite long time if the user-space would give a huge buffer, and this is a likely cause of some weird behavior spotted by syzcaller fuzzer. This patch puts a simple workaround, just adding a mutex break in the loop when a large number of events have been processed. This shouldn't hit any performance drop because the threshold is set high enough for usual operations. Fixes: 7bd800915677 ("ALSA: seq: More protection for concurrent write and ioctl races") Reported-by: syzbot+97aae04ce27e39cbfca9@syzkaller.appspotmail.com Reported-by: syzbot+4c595632b98bb8ffcc66@syzkaller.appspotmail.com Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: dapm: Adapt for debugfs API changeMark Brown2019-07-261-8/+10
| | | | | | | | | | | | | | | | | commit ceaea851b9ea75f9ea2bbefb53ff0d4b27cd5a6e upstream. Back in ff9fb72bc07705c (debugfs: return error values, not NULL) the debugfs APIs were changed to return error pointers rather than NULL pointers on error, breaking the error checking in ASoC. Update the code to use IS_ERR() and log the codes that are returned as part of the error messages. Fixes: ff9fb72bc07705c (debugfs: return error values, not NULL) Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: Intel: hdac_hdmi: Set ops to NULL on removeAmadeusz Sławiński2019-07-261-0/+6
| | | | | | | | | | | | | | [ Upstream commit 0f6ff78540bd1b4df1e0f17806b0ce2e1dff0d78 ] When we unload Skylake driver we may end up calling hdac_component_master_unbind(), it uses acomp->audio_ops, which we set in hdmi_codec_probe(), so we need to set it to NULL in hdmi_codec_remove(), otherwise we will dereference no longer existing pointer. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: meson: axg-tdm: fix sample clock inversionJerome Brunet2019-07-261-1/+1
| | | | | | | | | | | | | | | | [ Upstream commit cb36ff785e868992e96e8b9e5a0c2822b680a9e2 ] The content of SND_SOC_DAIFMT_FORMAT_MASK is a number, not a bitfield, so the test to check if the format is i2s is wrong. Because of this the clock setting may be wrong. For example, the sample clock gets inverted in DSP B mode, when it should not. Fix the lrclk invert helper function Fixes: 1a11d88f499c ("ASoC: meson: add tdm formatter base driver") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/realtek - Headphone Mic can't record after S3Kailang Yang2019-07-141-1/+1
| | | | | | | | | | | | | | | | | commit d07a9a4f66e944fcc900812cbc2f6817bde6a43d upstream. Dell headset mode platform with ALC236. It doesn't recording after system resume from S3. S3 mode was deep. s2idle was not has this issue. S3 deep will cut of codec power. So, the register will back to default after resume back. This patch will solve this issue. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb-audio: Fix parse of UAC2 Extension UnitsTakashi Iwai2019-07-141-6/+10
| | | | | | | | | | | | | | | | | | | | | | | commit ca95c7bf3d29716916baccdc77c3c2284b703069 upstream. Extension Unit (XU) is used to have a compatible layout with Processing Unit (PU) on UAC1, and the usb-audio driver code assumed it for parsing the descriptors. Meanwhile, on UAC2, XU became slightly incompatible with PU; namely, XU has a one-byte bmControls bitmap while PU has two bytes bmControls bitmap. This incompatibility results in the read of a wrong address for the last iExtension field, which ended up with an incorrect string for the mixer element name, as recently reported for Focusrite Scarlett 18i20 device. This patch corrects this misalignment by introducing a couple of new macros and calling them depending on the descriptor type. Fixes: 23caaf19b11e ("ALSA: usb-mixer: Add support for Audio Class v2.0") Reported-by: Stefan Sauer <ensonic@hora-obscura.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda: Initialize power_state field properlyTakashi Iwai2019-07-101-0/+1
| | | | | | | | | | | | | | | [ Upstream commit 183ab39eb0ea9879bb68422a83e65f750f3192f0 ] The recent commit 98081ca62cba ("ALSA: hda - Record the current power state before suspend/resume calls") made the HD-audio driver to store the PM state in power_state field. This forgot, however, the initialization at power up. Although the codec drivers usually don't need to refer to this field in the normal operation, let's initialize it properly for consistency. Fixes: 98081ca62cba ("ALSA: hda - Record the current power state before suspend/resume calls") Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/realtek - Change front mic location for Lenovo M710qDennis Wassenberg2019-07-101-0/+1
| | | | | | | | | | | | | commit bef33e19203dde434bcdf21c449e3fb4f06c2618 upstream. On M710q Lenovo ThinkCentre machine, there are two front mics, we change the location for one of them to avoid conflicts. Signed-off-by: Dennis Wassenberg <dennis.wassenberg@secunet.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Add quirks for several Clevo notebook barebonesRichard Sailer2019-07-101-3/+4
| | | | | | | | | | | | | | | | | | | | | | commit 503d90b30602a3295978e46d844ccc8167400fe6 upstream. This adds 4 SND_PCI_QUIRK(...) lines for several barebone models of the ODM Clevo. The model names are written in regex syntax to describe/match all clevo models that are similar enough and use the same PCI SSID that this fixup works for them. Additionally the lines regarding SSID 0x96e1 and 0x97e1 didn't fix audio for the all our Clevo notebooks using these SSIDs (models Clevo P960* and P970*) since ALC1220_FIXP_CLEVO_PB51ED_PINS swapped pins that are not necesarry to be swapped. This patch initiates ALC1220_FIXUP_CLEVO_P950 instead for these model and fixes the audio. Fixes: 80690a276f44 ("ALSA: hda/realtek - Add quirk for Tuxedo XC 1509") Signed-off-by: Richard Sailer <rs@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-19 08:31:31 +0000