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* ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()Kuninori Morimoto2019-04-051-3/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit a0c426fe143328760c9fd565cd203a37a7b4fde8 ] We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: dice: add support for Solid State Logic Duende Classic/MiniTakashi Sakamoto2019-04-051-1/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit b2e9e1c8810ee05c95f4d55800b8afae70ab01b4 ] Duende Classic was produced by Solid State Logic in 2006, as a first model of Duende DSP series. The following model, Duende Mini was produced in 2008. They are designed to receive isochronous packets for PCM frames via IEEE 1394 bus, perform signal processing by downloaded program, then transfer isochronous packets for converted PCM frames. These two models includes the same embedded board, consists of several ICs below: - Texus Instruments Inc, TSB41AB3 for physical layer of IEEE 1394 bus - WaveFront semiconductor, DICE II STD ASIC for link/protocol layer - Altera MAX 3000A CPLD for programs - Analog devices, SHARC ADSP-21363 for signal processing (4 chips) This commit adds support for the two models to ALSA dice driver. Like support for the other devices, packet streaming is just available. Userspace applications should be developed if full features became available; e.g. program uploader and parameter controller. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': False, 'chip_ID': 349771402425, 'cmc': True, 'cyc_clk_acc': 255, 'generation': 1, 'imc': True, 'isc': True, 'link_spd': 2, 'max_ROM': 1, 'max_rec': 512, 'name': '1394', 'node_vendor_ID': 20674, 'pmc': False}, 'root-directory': [ ['VENDOR', 20674], ['DESCRIPTOR', 'Solid State Logic'], ['MODEL', 112], ['DESCRIPTOR', 'Duende board'], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': True}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], [ 'UNIT', [ ['SPECIFIER_ID', 20674], ['VERSION', 1], ['MODEL', 112], ['DESCRIPTOR', 'Duende board']]]]} Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probewen yang2019-04-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010@hotmil.com> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: PCM: check if ops are defined before suspending PCMRanjani Sridharan2019-04-051-0/+8
| | | | | | | | | | | | | | | | | | | | | [ Upstream commit d9c0b2afe820fa3b3f8258a659daee2cc71ca3ef ] BE dai links only have internal PCM's and their substream ops may not be set. Suspending these PCM's will result in their ops->trigger() being invoked and cause a kernel oops. So skip suspending PCM's if their ops are NULL. [ NOTE: this change is required now for following the recent PCM core change to get rid of snd_pcm_suspend() call. Since DPCM BE takes the runtime carried from FE while keeping NULL ops, it can hit this bug. See details at: https://github.com/thesofproject/linux/pull/582 -- tiwai ] Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Sasha Levin <sashal@kernel.org>
* SoC: imx-sgtl5000: add missing put_device()Wen Yang2019-04-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010@hotmail.com> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: Shawn Guo <shawnguo@kernel.org> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Pengutronix Kernel Team <kernel@pengutronix.de> Cc: NXP Linux Team <linux-imx@nxp.com> Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-arm-kernel@lists.infradead.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ASoC: qcom: Fix of-node refcount unbalance in qcom_snd_parse_of()Takashi Iwai2019-04-051-2/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | [ Upstream commit 70b773219a32c7b8f3e53e041bc023ad99fd81f4 ] Although qcom_snd_parse_of() tries to manage the of-node refcount, there are still a few places that lead to the unblanced refcount in the error code path. Namely, - for_each_child_of_node() needs to unreference the iterator node if aborting the loop in the middle, - cpu, codec and platform node objects have to be unreferenced at each iteration, - platform and codec node objects have to be referred before jumping to the error handling code that unreference them unconditionally. This patch tries to address these by moving the assignment of platform and codec node objects to the beginning of the loop and adding the of_node_put() calls adequately. Fixes: c25e295cd77b ("ASoC: qcom: Add support to parse common audio device nodes") Cc: Patrick Lai <plai@codeaurora.org> Cc: Banajit Goswami <bgoswami@codeaurora.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
* ALSA: hda/realtek - Fix speakers on Acer Predator Helios 500 Ryzen laptopsBernhard Rosenkraenzer2019-04-031-0/+10
| | | | | | | | | | | | | | | | | | | | commit e2a829b3da01b9b32c4d0291d042b8a6e2a98ca3 upstream. On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers don't work out of the box. The problem can be worked around with hdajackretask, remapping the "Black Headphone, Right side" pin (0x21) to the Internal speaker. This patch adds a quirk to change this mapping by default. [ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the latest tree by tiwai ] Signed-off-by: Bernhard Rosenkraenzer <bero@lindev.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Enable headset MIC of ASUS X430UN and X512DK with ALC256Jian-Hong Pan2019-04-031-0/+4
| | | | | | | | | | | | | | commit 6ac371aa1a74240fb910c98aa3484d5ece8473d3 upstream. The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@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: Enable headset mic of ASUS P5440FF with ALC256Chris Chiu2019-04-031-0/+4
| | | | | | | | | | | | | | | commit a806ef1cf3bbc0baadc6cdeb11f12b5dd27e91c2 upstream. The ASUS laptop P5440FF with ALC256 can't detect the headset microphone until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@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: Enable ASUS X441MB and X705FD headset MIC with ALC256Jian-Hong Pan2019-04-031-0/+14
| | | | | | | | | | | | | | | commit e1037354a0a75acdea2b27043c0a371ed85cf262 upstream. The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@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 support for Acer Aspire E5-523G/ES1-432 headset micChris Chiu2019-04-031-0/+2
| | | | | | | | | | | | | | | | commit c7531e31c8a440b5fe6bd62664def5bcb6262f96 upstream. The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu@endlessm.com> Signed-off-by: Daniel Drake <drake@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: Enable headset MIC of Acer Aspire Z24-890 with ALC286Jian-Hong Pan2019-04-031-0/+1
| | | | | | | | | | | | | | commit 2733ccebf4a937a0858e7d05a4a003b89715033f upstream. The Acer Aspire Z24-890 cannot detect the headset MIC until ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Daniel Drake <drake@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: Enable headset MIC of Acer AIO with ALC286Jian-Hong Pan2019-04-031-3/+14
| | | | | | | | | | | | | | | | | commit 667a8f73753908c4d0171e52b71774f9be5d6713 upstream. Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot record sound from headset MIC. This patch adds the ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue. Fixes: 9f8aefed9623 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G") Fixes: b72f936f6b32 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G") Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Reviewed-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: hda/realtek - Add support headset mode for New DELL WYSE NBKailang Yang2019-04-031-0/+1
| | | | | | | | | | | commit da484d00f020af3dd7cfcc6c4b69a7f856832883 upstream. Enable headset mode support for new WYSE NB platform. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Add support headset mode for DELL WYSE AIOKailang Yang2019-04-031-0/+26
| | | | | | | | | | | commit 136824efaab2c095fc911048f7c7ddeda258c965 upstream. This patch will enable WYSE AIO for Headset mode. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: merge alc_fixup_headset_jack to alc295_fixup_chromebookJaroslav Kysela2019-04-031-30/+13
| | | | | | | | | | | | | | | commit c8a9afa632f0fd45731d3353525faf1fdb362c89 upstream. The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook. There are no other users for ALC225_FIXUP_HEADSET_JACK other than the chromebook hardware. Fixes: 10f5b1b85ed1 ("ALSA: hda/realtek - Fixed Headset Mic JD not stable") Cc: Kailang Yang <kailang@realtek.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Fixed Headset Mic JD not stableKailang Yang2019-04-031-1/+28
| | | | | | | | | | | | | commit 10f5b1b85ed10a80d45bc2db450e65bd792efaad upstream. It will be lose Mic JD state when Chrome OS boot and headset was plugged. Implement of reset combo jack JD. It will show normally. Fixes: e854747d7593 ("ALSA: hda/realtek - Enable headset button support for new codec") Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: pcm: Don't suspend stream in unrecoverable PCM stateTakashi Iwai2019-04-031-1/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 113ce08109f8e3b091399e7cc32486df1cff48e7 upstream. Currently PCM core sets each opened stream forcibly to SUSPENDED state via snd_pcm_suspend_all() call, and the user-space is responsible for re-triggering the resume manually either via snd_pcm_resume() or prepare call. The scheme works fine usually, but there are corner cases where the stream can't be resumed by that call: the streams still in OPEN state before finishing hw_params. When they are suspended, user-space cannot perform resume or prepare because they haven't been set up yet. The only possible recovery is to re-open the device, which isn't nice at all. Similarly, when a stream is in DISCONNECTED state, it makes no sense to change it to SUSPENDED state. Ditto for in SETUP state; which you can re-prepare directly. So, this patch addresses these issues by filtering the PCM streams to be suspended by checking the PCM state. When a stream is in either OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend action is skipped. To be noted, this problem was originally reported for the PCM runtime PM on HD-audio. And, the runtime PM problem itself was already addressed (although not intended) by the code refactoring commits 3d21ef0b49f8 ("ALSA: pcm: Suspend streams globally via device type PM ops") and 17bc4815de58 ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls"). These commits eliminated the snd_pcm_suspend*() calls from the runtime PM suspend callback code path, hence the racy OPEN state won't appear while runtime PM. (FWIW, the race window is between snd_pcm_open_substream() and the first power up in azx_pcm_open().) Although the runtime PM issue was already "fixed", the same problem is still present for the system PM, hence this patch is still needed. And for stable trees, this patch alone should suffice for fixing the runtime PM problem, too. Reported-and-tested-by: Jon Hunter <jonathanh@nvidia.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: pcm: Fix possible OOB access in PCM oss pluginsTakashi Iwai2019-04-031-21/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit ca0214ee2802dd47239a4e39fb21c5b00ef61b22 upstream. The PCM OSS emulation converts and transfers the data on the fly via "plugins". The data is converted over the dynamically allocated buffer for each plugin, and recently syzkaller caught OOB in this flow. Although the bisection by syzbot pointed out to the commit 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer allocations"), this is merely a commit to replace vmalloc() with kvmalloc(), hence it can't be the cause. The further debug action revealed that this happens in the case where a slave PCM doesn't support only the stereo channels while the OSS stream is set up for a mono channel. Below is a brief explanation: At each OSS parameter change, the driver sets up the PCM hw_params again in snd_pcm_oss_change_params_lock(). This is also the place where plugins are created and local buffers are allocated. The problem is that the plugins are created before the final hw_params is determined. Namely, two snd_pcm_hw_param_near() calls for setting the period size and periods may influence on the final result of channels, rates, etc, too, while the current code has already created plugins beforehand with the premature values. So, the plugin believes that channels=1, while the actual I/O is with channels=2, which makes the driver reading/writing over the allocated buffer size. The fix is simply to move the plugin allocation code after the final hw_params call. Reported-by: syzbot+d4503ae45b65c5bc1194@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: seq: oss: Fix Spectre v1 vulnerabilityGustavo A. R. Silva2019-04-031-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | | commit c709f14f0616482b67f9fbcb965e1493a03ff30b upstream. dev is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap) Fix this by sanitizing dev before using it to index dp->synths. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: rawmidi: Fix potential Spectre v1 vulnerabilityGustavo A. R. Silva2019-04-031-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | commit 2b1d9c8f87235f593826b9cf46ec10247741fff9 upstream. info->stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap) Fix this by sanitizing info->stream before using it to index rmidi->streams. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Enforces runtime_resume after S3 and S4 for each codecHui Wang2019-03-271-3/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit b5a236c175b0d984552a5f7c9d35141024c2b261 upstream. Recently we found the audio jack detection stop working after suspend on many machines with Realtek codec. Sometimes the audio selection dialogue didn't show up after users plugged headhphone/headset into the headset jack, sometimes after uses plugged headphone/headset, then click the sound icon on the upper-right corner of gnome-desktop, it also showed the speaker rather than the headphone. The root cause is that before suspend, the codec already call the runtime_suspend since this codec is not used by any apps, then in resume, it will not call runtime_resume for this codec. But for some realtek codec (so far, alc236, alc255 and alc891) with the specific BIOS, if it doesn't run runtime_resume after suspend, all codec functions including jack detection stop working anymore. This problem existed for a long time, but it was not exposed, that is because when problem happens, if users play sound or open sound-setting to check audio device, this will trigger calling to runtime_resume (via snd_hda_power_up), then the codec starts working again before users notice this problem. Since we don't know how many codec and BIOS combinations have this problem, to fix it, let the driver call runtime_resume for all codecs in pm_resume, maybe for some codecs, this is not needed, but it is harmless. After a codec is runtime resumed, if it is not used by any apps, it will be runtime suspended soon and furthermore we don't run suspend frequently, this change will not add much power consumption. Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control") 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 - Record the current power state before suspend/resume callsTakashi Iwai2019-03-271-2/+41
| | | | | | | | | | | | | | | | | | | commit 98081ca62cbac31fb0f7efaf90b2e7384ce22257 upstream. Currently we deal with single codec and suspend codec callbacks for all S3, S4 and runtime PM handling. But it turned out that we want distinguish the call patterns sometimes, e.g. for applying some init sequence only at probing and restoring from hibernate. This patch slightly modifies the common PM callbacks for HD-audio codec and stores the currently processed PM event in power_state of the codec's device.power field, which is currently unused. The codec callback can take a look at this event value and judges which purpose it's being called. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: ac97: Fix of-node refcount unbalanceTakashi Iwai2019-03-271-1/+1
| | | | | | | | | | | | | | | | commit 31d2350d602511efc9ef626b848fe521233b0387 upstream. ac97_of_get_child_device() take the refcount of the node explicitly via of_node_get(), but this leads to an unbalance. The for_each_child_of_node() loop itself takes the refcount for each iteration node, hence you don't need to take the extra refcount again. Fixes: 2225a3e6af78 ("ALSA: ac97: add codecs devicetree binding") Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Don't trigger jackpoll_work in azx_resumeHui Wang2019-03-271-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | commit 744c67ffeb06f2d2493f4049ba0bd19698ce0adf upstream. The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed the behaviour of azx_resume(), it triggers the jackpoll_work after applying this commit. This change introduced a new issue, all codecs are runtime active after S3, and will not call runtime_suspend() automatically. The root cause is the jackpoll_work calls snd_hda_power_up/down_pm, and it calls up_pm before snd_hdac_enter_pm is called, while calls the down_pm in the middle of enter_pm and leave_pm is called. This makes the dev->power.usage_count unbalanced after S3. To fix it, let azx_resume() don't trigger jackpoll_work as before it did. Fixes: 3baffc4a84d7 ("ALSA: hda/intel: Refactoring PM code") 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: firewire-motu: use 'version' field of unit directory to identify modelTakashi Sakamoto2019-03-271-10/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 2d012c65a9ca26a0ef87ea0a42f1653dd37155f5 upstream. Current ALSA firewire-motu driver uses the value of 'model' field of unit directory in configuration ROM for modalias for MOTU FireWire models. However, as long as I checked, Pre8 and 828mk3(Hybrid) have the same value for the field (=0x100800). unit | version | model --------------- | --------- | ---------- 828mkII | 0x000003 | 0x101800 Traveler | 0x000009 | 0x107800 Pre8 | 0x00000f | 0x100800 <- 828mk3(FW) | 0x000015 | 0x106800 AudioExpress | 0x000033 | 0x104800 828mk3(Hybrid) | 0x000035 | 0x100800 <- When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3, I got change of the value from 0x100800 to 0x103800. On the other hand, the value of 'version' field is fixed to 0x00000f. As a quick glance, the higher 12 bits of the value of 'version' field represent firmware version, while the lower 12 bits is unknown. By induction, the value of 'version' field represents actual model. This commit changes modalias to match the value of 'version' field, instead of 'model' field. For degug, long name of added sound card includes hexadecimal value of 'model' field. Fixes: 6c5e1ac0e144 ("ALSA: firewire-motu: add support for Motu Traveler") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # v4.19+ Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - add Lenovo IdeaCentre B550 to the power_save_blacklistJaroslav Kysela2019-03-271-2/+4
| | | | | | | | | | | | | | | | commit 721f1e6c1fd137e7e2053d8e103b666faaa2d50c upstream. Another machine which does not like the power saving (noise): https://bugzilla.redhat.com/show_bug.cgi?id=1689623 Also, reorder the Lenovo C50 entry to keep the table sorted. Reported-by: hs.guimaraes@outlook.com Signed-off-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: pcm186x: Fix energysense SLEEP bitCodrin Ciubotariu2019-03-231-3/+3
| | | | | | | | | | | | | | | commit 05bd7fcdd06b19a10f069af1bea3ad9abac038d7 upstream. The ADCs are sleeping when the SLEEP bit is set and running when it's cleared, so the bit should be inverted. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Acked-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_SCALE()Codrin Ciubotariu2019-03-231-1/+1
| | | | | | | | | | | | | | commit fcf4daabf08079e6d09958a2992e7446ef8d0438 upstream. According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Acked-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: fsl_esai: fix register setting issue in RIGHT_J modeS.j. Wang2019-03-231-3/+4
| | | | | | | | | | | | | | | | commit cc29ea007347f39f4c5a4d27b0b555955a0277f9 upstream. The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable@vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Reviewed-by: Fabio Estevam <festevam@gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek: Enable headset MIC of Acer TravelMate X514-51T with ALC255Jian-Hong Pan2019-03-191-0/+12
| | | | | | | | | | | | | | | | commit cbc05fd6708c1744ee6a61cb4c461ff956d30524 upstream. The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the internal DMIC uses another module - snd_soc_skl as the driver. We still need the NID 0x1a in the quirk to enable the headset MIC. Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> 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: hda/realtek - Reduce click noise on Dell Precision 5820 headphoneTakashi Iwai2019-03-191-23/+34
| | | | | | | | | | | | | | | | | | | | | | | | commit c0ca5eced22215c1e03e3ad479f8fab0bbb30772 upstream. Dell Precision 5820 with ALC3234 codec (which is equivalent with ALC255) shows click noises at (runtime) PM resume on the headphone. The biggest source of the noise comes from the cleared headphone pin control at resume, which is done via the standard shutup procedure. Although we have an override of the standard shutup callback to replace with NOP, this would skip other needed stuff (e.g. the pull down of headset power). So, instead, this "fixes" the behavior of alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag. When this flag is set, Realtek codec won't call the standard snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this flag instead of overriding spec->shutup callback itself. This allows us to apply the similar fix for other entries easily if needed in future. 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 audio jacks of ASUS UX362FA with ALC294Jian-Hong Pan2019-03-191-1/+4
| | | | | | | | | | | | | | | | | | | | | commit 8bb37a2a4d7c02affef554f5dc05f6d2e39c31f9 upstream. The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs through the internal speaker and the headphone. This issue can be fixed by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294. Besides, ASUS UX362FA and UX533FD have the same audio initial pin config values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD. Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294") Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com> Signed-off-by: Ming Shuo Chiu <chiu@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 - add more quirks for HP Z2 G4 and HP Z240Jaroslav Kysela2019-03-192-2/+7
| | | | | | | | | | | | | | commit 167897f4b32c2bc18b3b6183029a33fb420a114e upstream. Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and HP Z240 models. Reported-by: Jeff Burrell <jeff.burrell@hp.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda: Extend i915 component bind timeoutTakashi Iwai2019-03-191-2/+2
| | | | | | | | | | | | | | | | | | | | | commit cfc35f9c128cea8fce6a5513b1de50d36f3b209f upstream. I set 10 seconds for the timeout of the i915 audio component binding with a hope that recent machines are fast enough to handle all probe tasks in that period, but I was too optimistic. The binding may take longer than that, and this caused a problem on the machine with both audio and graphics driver modules loaded in parallel, as Paul Menzel experienced. This problem haven't hit so often just because the KMS driver is loaded in initrd on most machines. As a simple workaround, extend the timeout to 60 seconds. Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: firewire-motu: fix construction of PCM frame for capture directionTakashi Sakamoto2019-03-191-1/+3
| | | | | | | | | | | | | | | | | | | | | | | commit f97a0944a72b26a2bece72516294e112a890f98a upstream. In data blocks of common isochronous packet for MOTU devices, PCM frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described in IEC 61883-6. The frames are not aligned to quadlet. For capture PCM substream, ALSA firewire-motu driver constructs PCM frames by reading data blocks byte-by-byte. However this operation includes bug for lower byte of the PCM sample. This brings invalid content of the PCM samples. This commit fixes the bug. Reported-by: Peter Sjöberg <autopeter@gmail.com> Cc: <stable@vger.kernel.org> # v4.12+ Fixes: 4641c9394010 ("ALSA: firewire-motu: add MOTU specific protocol layer") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against ↵Takashi Sakamoto2019-03-191-1/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Liquid Saffire 56 commit 7dc661bd8d3261053b69e4e2d0050cd1ee540fc1 upstream. ALSA bebob driver has an entry for Focusrite Saffire Pro 10 I/O. The entry matches vendor_id in root directory and model_id in unit directory of configuration ROM for IEEE 1394 bus. On the other hand, configuration ROM of Focusrite Liquid Saffire 56 has the same vendor_id and model_id. This device is an application of TCAT Dice (TCD2220 a.k.a Dice Jr.) however ALSA bebob driver can be bound to it randomly instead of ALSA dice driver. At present, drivers in ALSA firewire stack can not handle this situation appropriately. This commit uses more identical mod_alias for Focusrite Saffire Pro 10 I/O in ALSA bebob driver. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 042a829d bus_info_length 4, crc_length 42, crc 33437 404 31333934 bus_name "1394" 408 f0649222 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100, max_rec 9 (1024), max_rom 2, gen 2, spd 2 (S400) 40c 00130e01 company_id 00130e | 410 000606e0 device_id 01000606e0 | EUI-64 00130e01000606e0 root directory ----------------------------------------------------------------- 414 0009d31c directory_length 9, crc 54044 418 04000014 hardware version 41c 0c0083c0 node capabilities per IEEE 1394 420 0300130e vendor 424 81000012 --> descriptor leaf at 46c 428 17000006 model 42c 81000016 --> descriptor leaf at 484 430 130120c2 version 434 d1000002 --> unit directory at 43c 438 d4000006 --> dependent info directory at 450 unit directory at 43c ----------------------------------------------------------------- 43c 0004707c directory_length 4, crc 28796 440 1200a02d specifier id: 1394 TA 444 13010001 version: AV/C 448 17000006 model 44c 81000013 --> descriptor leaf at 498 dependent info directory at 450 ----------------------------------------------------------------- 450 000637c7 directory_length 6, crc 14279 454 120007f5 specifier id 458 13000001 version 45c 3affffc7 (immediate value) 460 3b100000 (immediate value) 464 3cffffc7 (immediate value) 468 3d600000 (immediate value) descriptor leaf at 46c ----------------------------------------------------------------- 46c 00056f3b leaf_length 5, crc 28475 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 466f6375 "Focu" 47c 73726974 "srit" 480 65000000 "e" descriptor leaf at 484 ----------------------------------------------------------------- 484 0004a165 leaf_length 4, crc 41317 488 00000000 textual descriptor 48c 00000000 minimal ASCII 490 50726f31 "Pro1" 494 30494f00 "0IO" descriptor leaf at 498 ----------------------------------------------------------------- 498 0004a165 leaf_length 4, crc 41317 49c 00000000 textual descriptor 4a0 00000000 minimal ASCII 4a4 50726f31 "Pro1" 4a8 30494f00 "0IO" $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 040442e4 bus_info_length 4, crc_length 4, crc 17124 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 00130e04 company_id 00130e | 410 018001e9 device_id 04018001e9 | EUI-64 00130e04018001e9 root directory ----------------------------------------------------------------- 414 00065612 directory_length 6, crc 22034 418 0300130e vendor 41c 8100000a --> descriptor leaf at 444 420 17000006 model 424 8100000e --> descriptor leaf at 45c 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000418a0 directory_length 4, crc 6304 434 1200130e specifier id 438 13000001 version 43c 17000006 model 440 8100000f --> descriptor leaf at 47c descriptor leaf at 444 ----------------------------------------------------------------- 444 00056f3b leaf_length 5, crc 28475 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 466f6375 "Focu" 454 73726974 "srit" 458 65000000 "e" descriptor leaf at 45c ----------------------------------------------------------------- 45c 000762c6 leaf_length 7, crc 25286 460 00000000 textual descriptor 464 00000000 minimal ASCII 468 4c495155 "LIQU" 46c 49445f53 "ID_S" 470 41464649 "AFFI" 474 52455f35 "RE_5" 478 36000000 "6" descriptor leaf at 47c ----------------------------------------------------------------- 47c 000762c6 leaf_length 7, crc 25286 480 00000000 textual descriptor 484 00000000 minimal ASCII 488 4c495155 "LIQU" 48c 49445f53 "ID_S" 490 41464649 "AFFI" 494 52455f35 "RE_5" 498 36000000 "6" Cc: <stable@vger.kernel.org> # v3.16+ Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* Merge tag 'sound-5.0' of ↵Linus Torvalds2019-02-204-4/+56
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Here are a few last-minute fixes for 5.0. The most significant one is the OF-node refcount fix for ASoC simple-card, which could be triggered on many boards. Another fix for ASoC core is for the error handling in topology, while others are device-specific fixes for Samsung and HD-audio" * tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ASoC: simple-card: fixup refcount_t underflow ASoC: topology: free created components in tplg load error ALSA: hda/realtek: Disable PC beep in passthrough on alc285 ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5 ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI
| * Merge tag 'asoc-fix-v5.0-rc6' of ↵Takashi Iwai2019-02-193-3/+15
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.0 A few small fixes, a driver fix for Samsung, a fix for refcounting of of_nodes in the simple-card driver that triggered on a lot of systems and a fix for topology error handling.
| | * ASoC: simple-card: fixup refcount_t underflowKuninori Morimoto2019-02-181-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-card and simple-scu-card. Then it had refcount underflow bug. This patch fixup it. We will get below error without this patch. OF: ERROR: Bad of_node_put() on /sound CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514 Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x150 show_stack+0x24/0x30 dump_stack+0xb0/0xec of_node_release+0xd0/0xd8 kobject_put+0x74/0xe8 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x40/0x68 asoc_simple_card_probe+0x604/0x730 platform_drv_probe+0x58/0xa8 ... Reported-by: Vicente Bergas <vicencb@gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: topology: free created components in tplg load errorBard liao2019-02-181-1/+7
| | | | | | | | | | | | | | | | | | | | | Topology resources are no longer needed if any element failed to load. Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: samsung: i2s: Fix prescaler setting for the secondary DAISylwester Nawrocki2019-02-121-1/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | Make sure i2s->rclk_srcrate is properly initialized also during playback through the secondary DAI. Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Acked-by: Krzysztof Kozlowski <krzk@kernel.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| * | ALSA: hda/realtek: Disable PC beep in passthrough on alc285Hui Wang2019-02-141-1/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It is reported that there's a constant background "hum/whitenoise" in the headset on the Lenovo X1 machines with the codec alc285, and it is confirmed that if we run the command below, the noise will stop. sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0 Then I consulted this issue with Kailang, he told me the pin 0x1d on this codec is used for PC beep in, the noise probably comes from this pin and we can also disable the PC beep in passthrough, then the PC beep in will not affect other sound playback. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581 Cc: <stable@vger.kernel.org> Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda/realtek - Headset microphone and internal speaker support for ↵Jeremy Soller2019-02-141-0/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | System76 oryp5 On the System76 Oryx Pro (oryp5), there is a headset microphone input attached to 0x19 that does not have a jack detect. In order to get it working, the pin configuration needs to be set correctly, and the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we have a separate microphone jack that is already configured correctly. Since the ALC1220 does not have a fixup similar to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the ALC269 in a way that it can be accessed from the alc1220_fixup_system76_oryp5 function. In addition, the alc1220_fixup_clevo_p950 needs to be applied to gain speaker output. Signed-off-by: Jeremy Soller <jeremy@system76.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge tag 'sound-5.0-rc7' of ↵Linus Torvalds2019-02-1212-32/+64
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "It's a bit of surprising that we've got more changes than hoped at this late stage, but they all don't look too scary but small fixes. One change in ALSA core side is again the PCM regression fix that was partially addressed for OSS, but now the all relevant change is reverted instead. Also, a few ASoC core fixes for UAF and OOB are included, while the rest are usual random device-specific fixes" * tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: pcm: Revert capture stream behavior change in blocking mode ALSA: usb-audio: Fix implicit fb endpoint setup by quirk ALSA: hda - Add quirk for HP EliteBook 840 G5 ASoC: samsung: Prevent clk_get_rate() calls in atomic context ASoC: rsnd: ssiu: correct shift bit for ssiu9 ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables ASoC: topology: fix oops/use-after-free case with dai driver ASoC: rsnd: fixup MIX kctrl registration ASoC: core: Allow soc_find_component lookups to match parent of_node ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter ASoC: MAINTAINERS: fsl: Change Fabio's email address ASoC: hdmi-codec: fix oops on re-probe
| * | ALSA: pcm: Revert capture stream behavior change in blocking modeTakashi Iwai2019-02-081-16/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the commit 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture"), we changed the behavior of __snd_pcm_lib_xfer() to return immediately with 0 when a capture stream has a high start_threshold. This was intended to be a correction of the behavior consistency and looked harmless, but this was the culprit of the recent breakage reported by syzkaller, which was fixed by the commit e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream"). At the time for the OSS fix, I didn't touch the behavior for ALSA native API, as assuming that this behavior actually is good. But this turned out to be also broken actually for a similar deployment, e.g. one thread goes to a write loop in blocking mode while another thread controls the start/stop of the stream manually. Overall, the original commit is harmful, and it brings less merit to keep that behavior. Let's revert it. Fixes: 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture") Fixes: e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream") Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge tag 'asoc-fix-v5.0-rc5' of ↵Takashi Iwai2019-02-089-15/+51
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v5.0 A selection of driver specific fixes here, along with a few core fixes: - A fixup for some MFD devices that were broken by the previous fixes for deferred probe. - A fix for potential out of bounds array accesses when ordering DAPM power/up down sequences. - Avoid use after free issue when unloading and reloading drivers using topologies.
| | * ASoC: samsung: Prevent clk_get_rate() calls in atomic contextSylwester Nawrocki2019-02-071-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch moves clk_get_rate() call from trigger() to hw_params() callback to avoid calling sleeping clk API from atomic context and prevent deadlock as indicated below. Before this change clk_get_rate() was being called with same spinlock held as the one passed to the clk API when registering clocks exposed by the I2S driver. [ 82.109780] BUG: sleeping function called from invalid context at kernel/locking/mutex.c:908 [ 82.117009] in_atomic(): 1, irqs_disabled(): 128, pid: 1554, name: speaker-test [ 82.124235] 3 locks held by speaker-test/1554: [ 82.128653] #0: cc8c5328 (snd_pcm_link_rwlock){...-}, at: snd_pcm_stream_lock_irq+0x20/0x38 [ 82.137058] #1: ec9eda17 (&(&substream->self_group.lock)->rlock){..-.}, at: snd_pcm_ioctl+0x900/0x1268 [ 82.146417] #2: 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4 [ 82.154650] irq event stamp: 8144 [ 82.157949] hardirqs last enabled at (8143): [<c0a0f574>] _raw_read_unlock_irq+0x24/0x5c [ 82.166089] hardirqs last disabled at (8144): [<c0a0f6a8>] _raw_read_lock_irq+0x18/0x58 [ 82.174063] softirqs last enabled at (8004): [<c01024e4>] __do_softirq+0x3a4/0x66c [ 82.181688] softirqs last disabled at (7997): [<c012d730>] irq_exit+0x140/0x168 [ 82.188964] Preemption disabled at: [ 82.188967] [<00000000>] (null) [ 82.195728] CPU: 6 PID: 1554 Comm: speaker-test Not tainted 5.0.0-rc5-00192-ga6e6caca8f03 #191 [ 82.204302] Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) [ 82.210376] [<c0111a54>] (unwind_backtrace) from [<c010d8f4>] (show_stack+0x10/0x14) [ 82.218084] [<c010d8f4>] (show_stack) from [<c09ef004>] (dump_stack+0x90/0xc8) [ 82.225278] [<c09ef004>] (dump_stack) from [<c0152980>] (___might_sleep+0x22c/0x2c8) [ 82.232990] [<c0152980>] (___might_sleep) from [<c0a0a2e4>] (__mutex_lock+0x28/0xa3c) [ 82.240788] [<c0a0a2e4>] (__mutex_lock) from [<c0a0ad80>] (mutex_lock_nested+0x1c/0x24) [ 82.248763] [<c0a0ad80>] (mutex_lock_nested) from [<c04923dc>] (clk_prepare_lock+0x78/0xec) [ 82.257079] [<c04923dc>] (clk_prepare_lock) from [<c049538c>] (clk_core_get_rate+0xc/0x5c) [ 82.265309] [<c049538c>] (clk_core_get_rate) from [<c0766b18>] (i2s_trigger+0x490/0x6d4) [ 82.273369] [<c0766b18>] (i2s_trigger) from [<c074fec4>] (soc_pcm_trigger+0x100/0x140) [ 82.281254] [<c074fec4>] (soc_pcm_trigger) from [<c07378a0>] (snd_pcm_do_start+0x2c/0x30) [ 82.289400] [<c07378a0>] (snd_pcm_do_start) from [<c07376cc>] (snd_pcm_action_single+0x38/0x78) [ 82.298065] [<c07376cc>] (snd_pcm_action_single) from [<c073a450>] (snd_pcm_ioctl+0x910/0x1268) [ 82.306734] [<c073a450>] (snd_pcm_ioctl) from [<c0292344>] (do_vfs_ioctl+0x90/0x9ec) [ 82.314443] [<c0292344>] (do_vfs_ioctl) from [<c0292cd4>] (ksys_ioctl+0x34/0x60) [ 82.321808] [<c0292cd4>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28) [ 82.329431] Exception stack(0xeb875fa8 to 0xeb875ff0) [ 82.334459] 5fa0: 00033c18 b6e31000 00000004 00004142 00033d80 00033d80 [ 82.342605] 5fc0: 00033c18 b6e31000 00008000 00000036 00008000 00000000 beea38a8 00008000 [ 82.350748] 5fe0: b6e3142c beea384c b6da9a30 b6c9212c [ 82.355789] [ 82.357245] ====================================================== [ 82.363397] WARNING: possible circular locking dependency detected [ 82.369551] 5.0.0-rc5-00192-ga6e6caca8f03 #191 Tainted: G W [ 82.376395] ------------------------------------------------------ [ 82.382548] speaker-test/1554 is trying to acquire lock: [ 82.387834] 6d2007f4 (prepare_lock){+.+.}, at: clk_prepare_lock+0x78/0xec [ 82.394593] [ 82.394593] but task is already holding lock: [ 82.400398] 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4 [ 82.408197] [ 82.408197] which lock already depends on the new lock. [ 82.416343] [ 82.416343] the existing dependency chain (in reverse order) is: [ 82.423795] [ 82.423795] -> #1 (&(&pri_dai->spinlock)->rlock){..-.}: [ 82.430472] clk_mux_set_parent+0x34/0xb8 [ 82.434975] clk_core_set_parent_nolock+0x1c4/0x52c [ 82.440347] clk_set_parent+0x38/0x6c [ 82.444509] of_clk_set_defaults+0xc8/0x308 [ 82.449186] of_clk_add_provider+0x84/0xd0 [ 82.453779] samsung_i2s_probe+0x408/0x5f8 [ 82.458376] platform_drv_probe+0x48/0x98 [ 82.462879] really_probe+0x224/0x3f4 [ 82.467037] driver_probe_device+0x70/0x1c4 [ 82.471716] bus_for_each_drv+0x44/0x8c [ 82.476049] __device_attach+0xa0/0x138 [ 82.480382] bus_probe_device+0x88/0x90 [ 82.484715] deferred_probe_work_func+0x6c/0xbc [ 82.489741] process_one_work+0x200/0x740 [ 82.494246] worker_thread+0x2c/0x4c8 [ 82.498408] kthread+0x128/0x164 [ 82.502131] ret_from_fork+0x14/0x20 [ 82.506204] (null) [ 82.508976] [ 82.508976] -> #0 (prepare_lock){+.+.}: [ 82.514264] __mutex_lock+0x60/0xa3c [ 82.518336] mutex_lock_nested+0x1c/0x24 [ 82.522756] clk_prepare_lock+0x78/0xec [ 82.527088] clk_core_get_rate+0xc/0x5c [ 82.531421] i2s_trigger+0x490/0x6d4 [ 82.535494] soc_pcm_trigger+0x100/0x140 [ 82.539913] snd_pcm_do_start+0x2c/0x30 [ 82.544246] snd_pcm_action_single+0x38/0x78 [ 82.549012] snd_pcm_ioctl+0x910/0x1268 [ 82.553345] do_vfs_ioctl+0x90/0x9ec [ 82.557417] ksys_ioctl+0x34/0x60 [ 82.561229] ret_fast_syscall+0x0/0x28 [ 82.565477] 0xbeea384c [ 82.568421] [ 82.568421] other info that might help us debug this: [ 82.568421] [ 82.576394] Possible unsafe locking scenario: [ 82.576394] [ 82.582285] CPU0 CPU1 [ 82.586792] ---- ---- [ 82.591297] lock(&(&pri_dai->spinlock)->rlock); [ 82.595977] lock(prepare_lock); [ 82.601782] lock(&(&pri_dai->spinlock)->rlock); [ 82.608975] lock(prepare_lock); [ 82.612268] [ 82.612268] *** DEADLOCK *** Fixes: 647d04f8e07a ("ASoC: samsung: i2s: Ensure the RCLK rate is properly determined") Reported-by: Krzysztof Kozłowski <krzk@kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: rsnd: ssiu: correct shift bit for ssiu9Jiada Wang2019-02-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently "0xf << 36" is used to clear SSIU-9 internal buffer state, which overflows 32-bit value according to user reference manual, it is always bit4 ~ bit7 of SSI_SYS_STATUS[1,3,5,7] registers indicate SSIU-9's buffer state, so "0xf << 4" should be used. This patch fix incorrect shifting issue in SSIU-9 case Fixes: commit b7169ddea2f2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg") Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count checkKuninori Morimoto2019-02-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
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