- 15 Oct, 2021 1 commit
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Takashi Sakamoto authored
Some of MOTU models allows software to configure their DSP parameters by accessing to their registers. The models multiplex messages for status of DSP into isochronous packet as well as PCM frames. The message includes information of hardware metering, MIDI message, current parameters of DSP. For my convenience, I call them as 'register DSP' model. This patch adds message parser for them to gather hardware meter information. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Link: https://lore.kernel.org/r/20211015080826.34847-2-o-takashi@sakamocchi.jpSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 14 Oct, 2021 7 commits
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Takashi Iwai authored
Both snd_pcm_delay() and snd_pcm_hwsync() do the almost same thing. The only difference is that the former calculate the delay, so unify them as a code cleanup, and treat NULL delay argument only for hwsync operation. Also, the patch does a slight code refactoring in snd_pcm_delay(). The initialization of the delay value is done in the caller side now. Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20211014145323.26506-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
So far we used to read the current value of the mixer element dynamically at the first access, and the error from a GET_CUR message is treated as a fatal error (unless QUIRK_IGNORE_CTL_ERROR is set). It's rather inconvenient, as most of GET_CUR errors are no fatal, and we can continue operation with assumption of some fixed value. This patch makes the USB-audio driver to change the behavior at probe time; now it tries to initialize the current value of each mixer element that is built from a feature unit (those for typically for mixer volumes and switches). When a read failure happens, it tries to set the known minimum value. After that point, a cached value is used always, hence we won't hit GET_CUR message error any longer. The error from GET_CUR message is still shown as a warning normally, but only once at the probe time, and it'll keep operating. If the message is confirmed to be harmless, it can be shut up by QUIRK_IGNORE_CTL_ERROR quirk flag, too. Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Link: https://lore.kernel.org/r/20211014130636.17860-4-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The error from snd_usb_lock_shutdown() indicates that the device is disconnected, hence it makes no sense to show any further control message error in get_ctl_value_v2(). Return the error directly instead. Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Link: https://lore.kernel.org/r/20211014130636.17860-3-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The error message in get_ctl_value_v2() (for UAC2/3) is shown via KERN_ERR level but it was intended to be rather a debug message as seen in get_ctl_value_v1() (for UAC1). This patch downgrade the printk level. Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Link: https://lore.kernel.org/r/20211014130636.17860-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
A back-merge of 5.15 branch into 5.16-devel branch for further development of USB and ALSA core stuff that depends on 5.15 fixes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Steven Clarkson authored
Apply existing PCI quirk to the Clevo PC50HS and related models to fix audio output on the built in speakers. Signed-off-by: Steven Clarkson <sc@lambdal.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211014133554.1326741-1-sc@lambdal.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Greg Kroah-Hartman authored
The Shciit Hel device responds to the ctl message for the mic capture switch with a timeout of -EPIPE: usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1 This seems safe to ignore as the device works properly with the control message quirk, so add it to the quirk table so all is good. Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linux-usb@vger.kernel.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 12 Oct, 2021 4 commits
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Jonas Hahnfeld authored
The device advertises 8 formats, but only a rate of 48kHz is honored by the hardware and 24 bits give chopped audio, so only report the one working combination. This fixes out-of-the-box audio experience with PipeWire which otherwise attempts to choose S24_3LE (while PulseAudio defaulted to S16_LE). Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211012200906.3492-1-hahnjo@hahnjo.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Kai Vehmanen authored
The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136 ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d1224 ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The recent support for the improved low-latency playback mode applied the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this was a slight overkill. The use of the flag above disables effectively both PCM status and control mmaps, while basically what we want to track is only about the appl_ptr update. For less restriction, use a more proper flag, SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control mmap. Fixes: d5f871f8 ("ALSA: usb-audio: Improved lowlatency playback support") Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Hui Wang authored
We need to define the codec pin 0x1b to be the mic, but somehow the mic doesn't support hot plugging detection, and Windows also has this issue, so we set it to phantom headset-mic. Also the determine_headset_type() often returns the omtp type by a mistake when we plug a ctia headset, this makes the mic can't record sound at all. Because most of the headset are ctia type nowadays and some machines have the fixed ctia type audio jack, it is possible this machine has the fixed ctia jack too. Here we set this mic jack to fixed ctia type, this could avoid the mic type detection mistake and make the ctia headset work stable. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214537Reported-and-tested-by: msd <msd.mmq@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Link: https://lore.kernel.org/r/20211012114748.5238-1-hui.wang@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 11 Oct, 2021 2 commits
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Takashi Iwai authored
Michael Forney reported an incorrect padding type that was defined in the commit 80fe7430 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control") for PCM control mmap data. His analysis is correct, and this caused the misplacements of PCM control data on 32bit arch and 32bit compat mode. The bug is that the __pad2 definition in __snd_pcm_mmap_control64 struct was wrongly with __pad_before_uframe, which should have been __pad_after_uframe instead. This struct is used in SYNC_PTR ioctl and control mmap. Basically this bug leads to two problems: - The offset of avail_min field becomes wrong, it's placed right after appl_ptr without padding on little-endian - When appl_ptr and avail_min are read as 64bit values in kernel side, the values become either zero or corrupted (mixed up) One good news is that, because both user-space and kernel misunderstand the wrong offset, at least, 32bit application running on 32bit kernel works as is. Also, 64bit applications are unaffected because the padding size is zero. The remaining problem is the 32bit compat mode; as mentioned in the above, avail_min is placed right after appl_ptr on little-endian archs, 64bit kernel reads bogus values for appl_ptr updates, which may lead to streaming bugs like jumping, XRUN or whatever unexpected. (However, we haven't heard any serious bug reports due to this over years, so practically seen, it's fairly safe to assume that the impact by this bug is limited.) Ideally speaking, we should correct the wrong mmap status control definition. But this would cause again incompatibility with the existing binaries, and fixing it (e.g. by renumbering ioctls) would be really messy. So, as of this patch, we only correct the behavior of 32bit compat mode and keep the rest as is. Namely, the SYNC_PTR ioctl is now handled differently in compat mode to read/write the 32bit values at the right offsets. The control mmap of 32bit apps on 64bit kernels has been already disabled (which is likely rather an overlook, but this worked fine at this time :), so covering SYNC_PTR ioctl should suffice as a fallback. Fixes: 80fe7430 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control") Reported-by: Michael Forney <mforney@mforney.org> Reviewed-by: Arnd Bergmann <arnd@arndb.de> Cc: <stable@vger.kernel.org> Cc: Rich Felker <dalias@libc.org> Link: https://lore.kernel.org/r/29QBMJU8DE71E.2YZSH8IHT5HMH@mforney.org Link: https://lore.kernel.org/r/20211010075546.23220-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Cameron Berkenpas authored
The previous patch's HDA verb initialization for the Lenovo 13s sequence was slightly off. This updated verb sequence has been tested and confirmed working. Fixes: ad7cc2d4 ("ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Cc: <stable@vger.kernel.org> Signed-off-by: Cameron Berkenpas <cam@neo-zeon.de> Link: https://lore.kernel.org/r/20211010225410.23423-1-cam@neo-zeon.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 10 Oct, 2021 1 commit
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William Overton authored
The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk@gmail.com> Link: https://lore.kernel.org/r/20211010145841.11907-1-willovertonuk@gmail.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 07 Oct, 2021 3 commits
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Takashi Iwai authored
When a stream is in the implicit feedback mode, it's more or less tied with a capture stream. Passing SNDRV_PCM_INFO_JOINT_DUPLEX may help for user-space to understand the situation. Link: https://lore.kernel.org/r/20211007083528.4184-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
In the case of hot-disconnection of a PCM device, all file operations except for close should be rejected. This patch adds more sanity checks in the file operation code paths. Link: https://lore.kernel.org/r/20211006142214.3089-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
It seems that a few recent AMD systems show the codec configuration errors at the early boot, while loading the driver at a later stage works magically. Although the root cause of the error isn't clear, it's certainly not bad to allow retrying the codec probe in such a case if that helps. This patch adds the capability for retrying the probe upon codec probe errors on the certain AMD platforms. The probe_work is changed to a delayed work, and at the secondary call, it'll jump to the codec probing. Note that, not only adding the re-probing, this includes the behavior changes in the codec configuration function. Namely, snd_hda_codec_configure() won't unregister the codec at errors any longer. Instead, its caller, azx_codec_configure() unregisters the codecs with the probe failures *if* any codec has been successfully configured. If all codec probe failed, it doesn't unregister but let it re-probed -- which is the most case we're seeing and this patch tries to improve. Even if the driver doesn't re-probe or give up, it will go to the "free-all" error path, hence the leftover codecs shall be disabled / deleted in anyway. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190801 Link: https://lore.kernel.org/r/20211006141940.2897-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 06 Oct, 2021 1 commit
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Werner Sembach authored
This applies a SND_PCI_QUIRK(...) to the TongFang PHxTxX1 barebone. This fixes the issue of the internal Microphone not working after booting another OS. When booting a certain another OS this barebone keeps some coeff settings even after a cold shutdown. These coeffs prevent the microphone detection from working in Linux, making the Laptop think that there is always an external microphone plugged-in and therefore preventing the use of the internal one. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211006130415.538243-1-wse@tuxedocomputers.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 05 Oct, 2021 2 commits
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Kailang Yang authored
In power save mode, the recording voice from headset mic will 2s more delay. Add this patch will solve this issue. [ minor coding style fix by tiwai ] Signed-off-by: Kailang Yang <kailang@realtek.com> Tested-by: Kai-Heng Feng <kai.heng.feng@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/ccb0cdd5bbd7486eabbd8d987d384cb0@realtek.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The Scarlett device series from Focusrite Novation seem requiring the sample rate validations as we've done for MOTU devices; otherwise the driver probes invalid audioformat entries that contain the sample rates that actually don't work, and this may result in an incomplete setup as reported recently. This patch adds the needed quirk flag for enabling the sample rate validation for Focusrite Novation devices. Fixes: fe773b87 ("ALSA: usb-audio: workaround for iface reset issue") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214493 Link: https://lore.kernel.org/r/20211004074050.28241-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 04 Oct, 2021 2 commits
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Werner Sembach authored
This applies a SND_PCI_QUIRK(...) to the Clevo X170KM-G barebone. This fixes the issue of the devices internal Speaker not working. Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211001133111.428249-3-wse@tuxedocomputers.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Werner Sembach authored
The string "Clevo X170" is not enough to unambiguously identify the correct device. Fixing it so another Clevo barebone name starting with "X170" can be added without causing confusion. Signed-off-by: Werner Sembach <wse@tuxedocomputers.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211001133111.428249-2-wse@tuxedocomputers.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 01 Oct, 2021 3 commits
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Takashi Iwai authored
The commit d215f63d ("ALSA: usb-audio: Check available frames for the next packet size") introduced the available frame size check, but the conversion forgot to initialize the temporary variable properly, and it resulted in a bogus calculation. This patch fixes it. Fixes: d215f63d ("ALSA: usb-audio: Check available frames for the next packet size") Reported-by: Colin Ian King <colin.king@canonical.com> Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com Link: https://lore.kernel.org/r/20211001105425.16191-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Chris Chiu authored
The headphone mic is not working on Dell Latitude laptops with ALC3254. The codec vendor id is 0x10ec0295 and share the same pincfg as defined in ALC295_STANDARD_PINS. So the ALC269_FIXUP_DELL1_MIC_NO_PRESENCE will be applied per alc269_pin_fixup_tbl[] but actually the headphone mic is using NID 0x1b instead of 0x1a. The ALC269_FIXUP_DELL4_MIC_NO_PRESENCE need to be applied instead. Use ALC269_FIXUP_DELL4_MIC_NO_PRESENCE for particular models before a generic fixup comes out. Signed-off-by: Chris Chiu <chris.chiu@canonical.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20211001062856.1037901-1-chris.chiu@canonical.comSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Geraldo Nascimento authored
Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc03 and subsequent 1a15718b were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com> Link: https://lore.kernel.org/r/YVYSnoQ7nxLXT0Dq@gedaySigned-off-by: Takashi Iwai <tiwai@suse.de>
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- 30 Sep, 2021 14 commits
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Takashi Iwai authored
John Keeping reported and posted a patch for a potential UAF in rawmidi sequencer destruction: the snd_rawmidi_dev_seq_free() may be called after the associated rawmidi object got already freed. After a deeper look, it turned out that the bug is rather the incorrect private_free call order for a snd_seq_device. The snd_seq_device private_free gets called at the release callback of the sequencer device object, while this was rather expected to be executed at the snd_device call chains that runs at the beginning of the whole card-free procedure. It's been broken since the rewrite of sequencer-device binding (although it hasn't surfaced because the sequencer device release happens usually right along with the card device release). This patch corrects the private_free call to be done in the right place, at snd_seq_device_dev_free(). Fixes: 7c37ae5c ("ALSA: seq: Rewrite sequencer device binding with standard bus") Reported-and-tested-by: John Keeping <john@metanate.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210930114114.8645-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
While draining a stream, ALSA PCM core stops the stream by issuing snd_pcm_stop() after all data has been sent out. And, at PCM trigger stop, currently USB-audio driver kills the in-flight URBs explicitly, then at sync-stop ops, sync with the finish of all remaining URBs. This might result in a drop of the drained samples as most of USB-audio devices / hosts allow relatively long in-flight samples (as a sort of FIFO). For avoiding the trimming, this patch changes the stream-stop behavior during PCM draining state. Under that condition, the pending URBs won't be killed. The leftover in-flight URBs are caught by the sync-stop operation that shall be performed after the trigger-stop operation. Link: https://lore.kernel.org/r/20210929080844.11583-10-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
This is another attempt to improve further the handling of playback stream in the low latency mode. The latest workaround in commit 4267c5a8 ("ALSA: usb-audio: Work around for XRUN with low latency playback") revealed that submitting URBs forcibly in advance may trigger XRUN easily. In the classical mode, this problem was avoided by practically delaying the submission of the actual data with the pre-submissions of silent data before triggering the stream start. But that is exactly what we want to avoid. Now, in this patch, instead of the previous workaround, we take a similar approach as used in the implicit feedback mode. The URBs are queued at the PCM trigger start like before, but we check whether the buffer has been already filled enough before each submission, and stop queuing if the data overcomes the threshold. The remaining URBs are kept in the ready list, and they will be retrieved in the URB complete callback of other (already queued) URBs. In the complete callback, we try to fill the data and submit as much as possible again. When there is no more available in-flight URBs that may handle the pending data, we'll check in PCM ack callback and submit and process URBs there in addition. In this way, the amount of in-flight URBs may vary dynamically and flexibly depending on the available data without hitting XRUN. The following things are changed to achieve the behavior above: * The endpoint prepare callback is changed to return an error code; when there is no enough data available, it may return -EAGAIN. Currently only prepare_playback_urb() returns the error. The evaluation of the available data is a bit messy here; we can't check with snd_pcm_avail() at the point of prepare callback (as runtime->status->hwptr hasn't been updated yet), hence we manually estimate the appl_ptr and compare with the internal hwptr_done to calculate the available frames. * snd_usb_endpoint_start() doesn't submit full URBs if the prepare callback returns -EAGAIN, and puts the remaining URBs to the ready list for the later submission. * snd_complete_urb() treats the URBs in the low-latency mode similarly like the implicit feedback mode, and submissions are done in (now exported) snd_usb_queue_pending_output_urbs(). * snd_usb_queue_pending_output_urbs() again checks the error value from the prepare callback. If it's -EAGAIN for the normal stream (i.e. not implicit feedback mode), we push it back to the ready list again. * PCM ack callback is introduced for the playback stream, and it calls snd_usb_queue_pending_output_urbs() if there is no in-flight URB while the stream is running. This corresponds to the case where the system needs the appl_ptr update for re-submitting a new URB. * snd_usb_queue_pending_output_urbs() and the prepare EP callback receive in_stream_lock argument, which is a bool flag indicating the call path from PCM ack. It's needed for avoiding the deadlock of snd_pcm_period_elapsed() calls. * Set the new SNDRV_PCM_INFO_EXPLICIT_SYNC flag when the new low-latency mode is deployed. This assures catching each applptr update even in the mmap mode. Fixes: 4267c5a8 ("ALSA: usb-audio: Work around for XRUN with low latency playback") Link: https://lore.kernel.org/r/20210929080844.11583-9-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
In theory, stop_urbs() may be called concurrently. Although we have the state check beforehand, it's safer to apply ep->lock during the critical list head manipulations. Link: https://lore.kernel.org/r/20210929080844.11583-8-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
This is yet more preparation for the upcoming changes. Extend snd_usb_endpoint_next_packet_size() to check the available frames and return -EAGAIN if the next packet size is equal or exceeds the given size. This will be needed for avoiding XRUN during the low latency operation. As of this patch, avail=0 is passed, i.e. the check is skipped and no behavior change. Link: https://lore.kernel.org/r/20210929080844.11583-7-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
When a playback stream runs in the implicit feedback mode, its operation is passive and won't start unless the capture packet is received. This behavior contradicts with the low-latency playback mode, and we should turn off lowlatency_playback flag accordingly. In theory, we may take the low-latency mode when the playback-first quirk is set, but it still conflicts with the later operation with the fixed packet numbers, so it's disabled all together for now. Link: https://lore.kernel.org/r/20210929080844.11583-6-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The free-wheel stream operation like dmix may not update the appl_ptr appropriately, and it doesn't fit with the low-latency playback mode. Disable the low-latency playback operation when the stream is set up in such a mode. Link: https://lore.kernel.org/r/20210929080844.11583-5-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
This is a preparation patch for the upcoming low-latency improvement changes. Rename early_playback_start flag with lowlatency_playback as it's more intuitive. The new flag is basically a reverse meaning. Along with the rename, factor out the code to set the flag to a function. This makes the complex condition checks simpler. Also, the same flag is introduced to snd_usb_endpoint, too, that is carried from the snd_usb_substream flag. Currently the endpoint flag isn't still referred, but will be used in later patches. Link: https://lore.kernel.org/r/20210929080844.11583-4-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
USB-audio driver tries to sync with the clear of all pending URBs in wait_clear_urbs(), and it waits for all bits in active_mask getting cleared. This works fine for the normal operations, but when a stream is managed in the implicit feedback mode, there is still a very thin race window: namely, in snd_complete_usb(), the active_mask bit for the current URB is once cleared before re-submitted in queue_pending_output_urbs(). If wait_clear_urbs() is called during that period, it may pass the test and go forward even though there may be a still pending URB. For covering it, this patch adds a new counter to each endpoint to keep the number of in-flight URBs, and changes wait_clear_urbs() checking this number instead. The counter is decremented at the end of URB complete, hence the reference is kept as long as the URB complete is in process. Link: https://lore.kernel.org/r/20210929080844.11583-3-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
When a single clock source is shared among several endpoints, we have to keep the same rate on all active endpoints as long as the clock is being used. For dealing with such a case, this patch adds one more check in the hw params constraint for the rate to take the shared clocks into account. The current rate is evaluated from the endpoint list that applies the same clock source. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190418 Link: https://lore.kernel.org/r/20210929080844.11583-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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John Liu authored
The Dell Precision 5560 laptop appears to use the 4-speakers-on-ALC289 audio just like its sibling product XPS 9510, so it requires the same quirk to enable woofer output. Tested on my Dell Precision 5560. Signed-off-by: John Liu <johnliu55tw@gmail.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20210930115316.659-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The commit f87e7f25 ("ALSA: hda - Improved position reporting on SKL+") changed the PCM position report for SKL+ chips to use DPIB, but according to Pierre, DPIB is no best choice for the accurate position reports and it often reports too early. The recommended method is rather the classical position buffer. This patch makes the PCM position reporting on SKL+ back to the position buffer again. Fixes: f87e7f25 ("ALSA: hda - Improved position reporting on SKL+") Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210929072934.6809-3-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The position reporting on Intel Skylake and later chips via azx_get_pos_skl() contains a udelay(20) call for the capture streams. A call for this alone doesn't sound too harmful. However, as the pointer PCM ops is one of the hottest path in the PCM operations -- especially for the timer-scheduled operations like PulseAudio -- such a delay hogs CPU usage significantly in the total performance. The code there was taken from the original code in ASoC SST Skylake driver blindly. The udelay() is a workaround for the case where the reported position is behind the period boundary at the timing triggered from interrupts; applications often expect that the full data is available for the whole period when returned (and also that's the definition of the ALSA PCM period). OTOH, HD-audio (legacy) driver has already some workarounds for the delayed position reporting due to its relatively large FIFO, such as the BDL position adjustment and the delayed period-elapsed call in the work. That said, the udelay() is almost superfluous for HD-audio driver unlike SST, and we can drop the udelay(). Though, the current code doesn't guarantee the full period readiness as mentioned in the above, but rather it checks the wallclock and detects the unexpected jump. That's one missing piece, and the drop of udelay() needs a bit more sanity checks for the delayed handling. This patch implements those: the drop of udelay() call in azx_get_pos_skl() and the more proper check of hwptr in azx_position_ok(). The latter change is applied only for the case where the stream is running in the normal mode without no_period_wakeup flag. When no_period_wakeup is set, it essentially ignores the period handling and rather concentrates only on the current position; which implies that we don't need to care about the period boundary at all. Fixes: f87e7f25 ("ALSA: hda - Improved position reporting on SKL+") Reported-by: Jens Axboe <axboe@kernel.dk> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20210929072934.6809-2-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The check of the returned error code is missing in scarlett2_update_monitor_other(). Let's fix it. Fixes: d5bda7e0 ("ALSA: usb-audio: scarlett2: Add support for the talkback feature") Reported-by: kernel test robot <lkp@intel.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/202109131831.9IodEzRx-lkp@intel.com Link: https://lore.kernel.org/r/20210929073540.9611-1-tiwai@suse.deSigned-off-by: Takashi Iwai <tiwai@suse.de>
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