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2020-11-23ALSA: usb-audio: Use unsigned char for iface and altsettings fieldsTakashi Iwai1-1/+1
Just for consistency, use unsigned char for iface and altsetting in allover places. Also rearrange the field positions of snd_usb_endpiont and tidy up with some comments. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-35-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Replace slave/master termsTakashi Iwai1-14/+14
Follow the inclusive terminology, just replace sync_master/sync_slave with sync_source/sync_sink. It's also a bit clearer from its meaning, too. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-34-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Unify the code for the next packet size calculationTakashi Iwai1-16/+24
There are two places calculating the next packet size for the playback stream in the exactly same way. Provide the single helper for this purpose and use it from both places gracefully. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-32-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Refactoring endpoint URB deactivationTakashi Iwai1-18/+23
Minor code refactoring to consolidate the URB deactivation code in endpoint.c. A slight behavior change is that the error handling in snd_usb_endpoint_start() leaves EP_FLAG_STOPPING now. This should be synced with the later PCM sync_stop callback. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-30-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Use atomic_t for endpoint use_countTakashi Iwai1-12/+14
The endpoint objects may be started/stopped concurrently by different substreams in the case of implicit feedback mode, while the current code handles the reference counter without any protection. This patch changes the refcount to atomic_t for avoiding the inconsistency. We need no reference_t here as the refcount goes only up to 2. Also the name "use_count" is renamed to "running" since this is about actually the running status, not the open refcount. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-29-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Constify audioformat pointer referencesTakashi Iwai1-1/+1
The audioformat is referred in many places but most of usages are read-only. Let's add const prefix in the possible places. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-28-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Fix possible stall of implicit fb packet ring-bufferTakashi Iwai1-23/+60
The implicit feedback mode uses a ring buffer for storing the received packet sizes from the feedback source, and the code has a slight flaw; when a playback stream stalls by some reason and the URBs aren't processed, the next_packet FIFO might become empty, but the driver can't distinguish whether it's empty or full because it's managed with read_poss and write_pos. This patch addresses those by changing the next_packet array management. Instead of keeping read and write positions, now the head position and the queued amount are kept. It's easier to understand about the emptiness. Also, the URB active flag is now cleared before calling queue_pending_output_urbs() for avoiding (theoretically) possible inconsistency. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-27-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Refactor endpoint managementTakashi Iwai1-326/+336
This is an intensive surgery for the endpoint and stream management for achieving more robust and clean code. The goals of this patch are: - More clear endpoint resource changes - The interface altsetting control in a single place Below are brief description of the whole changes. First off, most of the endpoint operations are moved into endpoint.c, so that the snd_usb_endpoint object is only referred in other places. The endpoint object is acquired and released via the new functions snd_usb_endpoint_open() and snd_usb_endpoint_close() that are called at PCM hw_params and hw_free callbacks, respectively. Those are ref-counted and EPs can manage the multiple opens. The open callback receives the audioformat and hw_params arguments, and those are used for initializing the EP parameters; especially the endpoint, interface and altset numbers are read from there, as well as the PCM parameters like the format, rate and channels. Those are stored in snd_usb_endpoint object. If it's the secondary open, the function checks whether the given parameters are compatible with the already opened EP setup, too. The coupling with a sync EP (including an implicit feedback sync) is done by the sole snd_usb_endpoint_set_sync() call. The configuration of each endpoint is done in a single shot via snd_usb_endpoint_configure() call. This is the place where most of PCM configurations are done. A few flags and special handling in the snd_usb_substream are dropped along with this change. A significant difference wrt the configuration from the previous code is the order of USB host interface setups. Now the interface is always disabled at beginning and (re-)enabled at the last step of snd_usb_endpoint_configure(), in order to be compliant with the standard UAC2/3. For UAC1, the interface is set before the parameter setups since there seem devices that require it (e.g. Yamaha THR10), just like how it was done in the previous driver code. The start/stop are almost same as before, also single-shots. The URB callbacks need to be set via snd_usb_endpoint_set_callback() like the previous code at the trigger phase, too. Finally, the flag for the re-setup is set at the device suspend through the full EP list, instead of PCM trigger. This catches the overlooked cases where the PCM hasn't been running yet but the device needs the full setup after resume. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-26-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Set callbacks via snd_usb_endpoint_set_callback()Takashi Iwai1-19/+41
The prepare_data_urb and retire_data_urb fields of the endpoint object are set dynamically at PCM trigger start/stop. Those are evaluated in the endpoint handler, but there can be a race, especially if two different PCM substreams are handling the same endpoint for the implicit feedback case. Also, the data_subs field of the endpoint is set and accessed dynamically, too, which has the same risk. As a slight improvement for the concurrency, this patch introduces the function to set the callbacks and the data in a shot with the memory barrier. In the reader side, it's also fetched with the memory barrier. There is still a room of race if prepare and retire callbacks are set during executing the URB completion. But such an inconsistency may happen only for the implicit fb source, i.e. it's only about the capture stream. And luckily, the capture stream never sets the prepare callback, hence the problem doesn't happen practically. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-23-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Stop both endpoints properly at errorTakashi Iwai1-0/+6
start_endpoints() may leave the data endpoint running if an error happens at starting the sync endpoint. We should stop both streams properly, instead. While we're at it, move the debug prints into the endpoint.c that is a more suitable place. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-22-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Create endpoint objects at parsing phaseTakashi Iwai1-53/+34
Currently snd_usb_endpoint objects are created at first when the substream is opened and tries to assign the endpoints corresponding to the matching audioformat. But since basically the all endpoints have been already parsed and the information have been obtained, we may create the endpoint objects statically at the init phase. It's easier to manage for the implicit fb case, for example. This patch changes the endpoint object management and lets the parser to create the all endpoint objects. This change shouldn't bring any functional changes. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-15-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Add hw constraint for implicit fb syncTakashi Iwai1-19/+53
In the current code, there is no check at the stream open time whether the endpoint is being already used by others. In the normal operations, this shouldn't happen, but in the case of the implicit feedback mode, it's a common problem with the full duplex operation, because the capture stream is always opened by the playback stream as an implicit sync source. Although we recently introduced the check of such a conflict of parameters at the PCM hw_params time, it doesn't give any hint at the hw_params itself and just gives the error. This isn't quite comfortable, and it caused problems on many applications. This patch attempts to make the parameter handling easier by introducing the strict hw constraint matching with the counterpart stream that is being used. That said, when an implicit feedback playback stream is running before a capture stream is opened, the capture stream carries the PCM hw-constraint to allow only the same sample rate, format, periods and period frames as the running playback stream. If not opened or there is no conflict of endpoints, the behavior remains as same as before. Note that this kind of "weak link" should work for most cases, but this is no concrete solution; e.g. if an application changes the hw params multiple times while another stream is opened, this would lead to inconsistencies. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-11-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Improve some debug printsTakashi Iwai1-4/+4
There are a few rooms for improvements wrt the debug prints: - The EP debug print is shown only at starting, not at stopping - The EP debug print contains useless object addresses - Some helpers show the urb and the EP object addresses, too This patch addresses those shortcomings. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-8-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-11-23ALSA: usb-audio: Add snd_usb_get_endpoint() helperTakashi Iwai1-9/+27
Factor out the code to obtain snd_usb_endpoint object matching with the given endpoint. It'll be used in the later patch to add the implicit feedback hw-constraint. No functional change by this patch itself. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-10-06ALSA: usb-audio: endpoint.c: fix repeated word 'there'Randy Dunlap1-1/+1
Drop the duplicated word "there". Signed-off-by: Randy Dunlap <rdunlap@infradead.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20201005191244.23902-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-08-06Merge tag 'sound-5.9-rc1' of ↵Linus Torvalds1-3/+2
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This became wide and scattered updates all over the sound tree as diffstat shows: lots of (still ongoing) refactoring works in ASoC, fixes and cleanups caught by static analysis, inclusive term conversions as well as lots of new drivers. Below are highlights: ASoC core: - API cleanups and conversions to the unified mute_stream() call - Simplify I/O helper functions - Use helper macros to retrieve RTD from substreams ASoC drivers: - Lots of fixes and cleanups in Intel ASoC drivers - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards, nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards, TI J721e EVM ALSA core: - Minor code refacotring for SG-buffer handling HD-audio: - Generalization of mute-LED handling with LED classdev - Intel silent stream support for HDMI - Device-specific fixes: CA0132, Loongson-3 Others: - Usual USB- and HD-audio quirks for various devices - Fixes for echoaudio DMA position handling - Various documents and trivial fixes for sparse warnings - Conversion to adopt inclusive terms" * tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits) ALSA: pci: delete repeated words in comments ALSA: isa: delete repeated words in comments ALSA: hda/tegra: Add 100us dma stop delay ALSA: hda: Add dma stop delay variable ASoC: hda/tegra: Set buffer alignment to 128 bytes ALSA: seq: oss: Serialize ioctls ALSA: hda/hdmi: Add quirk to force connectivity ALSA: usb-audio: add startech usb audio dock name ALSA: usb-audio: Add support for Lenovo ThinkStation P620 Revert "ALSA: hda: call runtime_allow() for all hda controllers" ALSA: hda/ca0132 - Fix AE-5 microphone selection commands. ALSA: hda/ca0132 - Add new quirk ID for Recon3D. ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value. ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops ALSA: docs: fix typo ALSA: doc: use correct config variable name ASoC: core: Two step component registration ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: sh: Replace 'select' DMADEVICES 'with depends on' ...
2020-08-04Merge tag 'uninit-macro-v5.9-rc1' of ↵Linus Torvalds1-1/+1
git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux Pull uninitialized_var() macro removal from Kees Cook: "This is long overdue, and has hidden too many bugs over the years. The series has several "by hand" fixes, and then a trivial treewide replacement. - Clean up non-trivial uses of uninitialized_var() - Update documentation and checkpatch for uninitialized_var() removal - Treewide removal of uninitialized_var()" * tag 'uninit-macro-v5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/kees/linux: compiler: Remove uninitialized_var() macro treewide: Remove uninitialized_var() usage checkpatch: Remove awareness of uninitialized_var() macro mm/debug_vm_pgtable: Remove uninitialized_var() usage f2fs: Eliminate usage of uninitialized_var() macro media: sur40: Remove uninitialized_var() usage KVM: PPC: Book3S PR: Remove uninitialized_var() usage clk: spear: Remove uninitialized_var() usage clk: st: Remove uninitialized_var() usage spi: davinci: Remove uninitialized_var() usage ide: Remove uninitialized_var() usage rtlwifi: rtl8192cu: Remove uninitialized_var() usage b43: Remove uninitialized_var() usage drbd: Remove uninitialized_var() usage x86/mm/numa: Remove uninitialized_var() usage docs: deprecated.rst: Add uninitialized_var()
2020-08-03Merge branch 'for-next' into for-linusTakashi Iwai1-3/+2
2020-07-27ALSA: usb-audio: endpoint : remove needless check before usb_free_coherent()Xu Wang1-3/+2
usb_free_coherent() is safe with NULL addr and this check is not required. Signed-off-by: Xu Wang <vulab@iscas.ac.cn> Link: https://lore.kernel.org/r/20200727025208.8739-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-16treewide: Remove uninitialized_var() usageKees Cook1-1/+1
Using uninitialized_var() is dangerous as it papers over real bugs[1] (or can in the future), and suppresses unrelated compiler warnings (e.g. "unused variable"). If the compiler thinks it is uninitialized, either simply initialize the variable or make compiler changes. In preparation for removing[2] the[3] macro[4], remove all remaining needless uses with the following script: git grep '\buninitialized_var\b' | cut -d: -f1 | sort -u | \ xargs perl -pi -e \ 's/\buninitialized_var\(([^\)]+)\)/\1/g; s:\s*/\* (GCC be quiet|to make compiler happy) \*/$::g;' drivers/video/fbdev/riva/riva_hw.c was manually tweaked to avoid pathological white-space. No outstanding warnings were found building allmodconfig with GCC 9.3.0 for x86_64, i386, arm64, arm, powerpc, powerpc64le, s390x, mips, sparc64, alpha, and m68k. [1] https://lore.kernel.org/lkml/20200603174714.192027-1-glider@google.com/ [2] https://lore.kernel.org/lkml/CA+55aFw+Vbj0i=1TGqCR5vQkCzWJ0QxK6CernOU6eedsudAixw@mail.gmail.com/ [3] https://lore.kernel.org/lkml/CA+55aFwgbgqhbp1fkxvRKEpzyR5J8n1vKT1VZdz9knmPuXhOeg@mail.gmail.com/ [4] https://lore.kernel.org/lkml/CA+55aFz2500WfbKXAx8s67wrm9=yVJu65TpLgN_ybYNv0VEOKA@mail.gmail.com/ Reviewed-by: Leon Romanovsky <leonro@mellanox.com> # drivers/infiniband and mlx4/mlx5 Acked-by: Jason Gunthorpe <jgg@mellanox.com> # IB Acked-by: Kalle Valo <kvalo@codeaurora.org> # wireless drivers Reviewed-by: Chao Yu <yuchao0@huawei.com> # erofs Signed-off-by: Kees Cook <keescook@chromium.org>
2020-06-30ALSA: usb-audio: Replace s/frame/packet/ where appropriateAlexander Tsoy1-10/+9
Replace several occurences of "frame" with a "packet" where appropriate. Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200629025934.154288-2-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-06-30ALSA: usb-audio: Fix packet size calculationAlexander Tsoy1-0/+1
Commit f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") introduced a regression for devices which have playback endpoints with bInterval > 1. Fix this by taking ep->datainterval into account. Note that frame and fps are actually mean packet and packets per second in the code introduces by the mentioned commit. This will be fixed in a follow-up patch. Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353 Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200629025934.154288-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-05-15ALSA: usb-audio: Add duplex sound support for USB devices using implicit ↵Erwin Burema1-4/+191
feedback For USB sound devices using implicit feedback the endpoint used for this feedback should be able to be opened twice, once for required feedback and second time for audio data. This way these devices can be put in duplex audio mode. Since this only works if the settings of the endpoint don't change a check is included for this. This fixes bug 207023 ("MOTU M2 regression on duplex audio") and should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device will not operate full-duplex") Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series") Signed-off-by: Erwin Burema <e.burema@gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751 Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24ALSA: usb-audio: Fix racy list management in output queueTakashi Iwai1-3/+3
The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-04-24ALSA: usb-audio: Improve frames size computationAlexander Tsoy1-5/+38
For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-11-13ALSA: usb-audio: not submit urb for stopped endpointHenry Lin1-0/+3
While output urb's snd_complete_urb() is executing, calling prepare_outbound_urb() may cause endpoint stopped before prepare_outbound_urb() returns and result in next urb submitted to stopped endpoint. usb-audio driver cannot re-use it afterwards as the urb is still hold by usb stack. This change checks EP_FLAG_RUNNING flag after prepare_outbound_urb() again to let snd_complete_urb() know the endpoint already stopped and does not submit next urb. Below kind of error will be fixed: [ 213.153103] usb 1-2: timeout: still 1 active urbs on EP #1 [ 213.164121] usb 1-2: cannot submit urb 0, error -16: unknown error Signed-off-by: Henry Lin <henryl@nvidia.com> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20191113021420.13377-1-henryl@nvidia.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-05-30treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 156Thomas Gleixner1-14/+1
Based on 1 normalized pattern(s): this program is free software you can redistribute it and or modify it under the terms of the gnu general public license as published by the free software foundation either version 2 of the license or at your option any later version this program is distributed in the hope that it will be useful but without any warranty without even the implied warranty of merchantability or fitness for a particular purpose see the gnu general public license for more details you should have received a copy of the gnu general public license along with this program if not write to the free software foundation inc 59 temple place suite 330 boston ma 02111 1307 usa extracted by the scancode license scanner the SPDX license identifier GPL-2.0-or-later has been chosen to replace the boilerplate/reference in 1334 file(s). Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Reviewed-by: Allison Randal <allison@lohutok.net> Reviewed-by: Richard Fontana <rfontana@redhat.com> Cc: linux-spdx@vger.kernel.org Link: https://lkml.kernel.org/r/20190527070033.113240726@linutronix.de Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2018-08-01ALSA: usb-audio: remove redundant pointer 'urb'Colin Ian King1-2/+0
Pointer 'urb' is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'urb' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-05ALSA: usb-audio: test EP_FLAG_RUNNING at urb completionIoan-Adrian Ratiu1-0/+3
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion logic allows us to save a few cpu cycles by returning early, skipping the pending urb in case the stream was stopped; the stop logic handles the urb and sets the completion callbacks to NULL. Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-05ALSA: usb-audio: Fix irq/process data synchronizationIoan-Adrian Ratiu1-10/+7
Commit 16200948d83 ("ALSA: usb-audio: Fix race at stopping the stream") was incomplete causing another more severe kernel panic, so it got reverted. This fixes both the original problem and its fallout kernel race/crash. The original fix is to move the endpoint member NULL clearing logic inside wait_clear_urbs() so the irq triggering the urb completion doesn't call retire_capture/playback_urb() after the NULL clearing and generate a panic. However this creates a new race between snd_usb_endpoint_start()'s call to wait_clear_urbs() and the irq urb completion handler which again calls retire_capture/playback_urb() leading to a new NULL dereference. We keep the EP deactivation code in snd_usb_endpoint_start() because removing it will break the EP reference counting (see [1] [2] for info), however we don't need the "can_sleep" mechanism anymore because a new function was introduced (snd_usb_endpoint_sync_pending_stop()) which synchronizes pending stops and gets called inside the pcm prepare callback. It also makes sense to remove can_sleep because it was also removed from deactivate_urbs() signature in [3] so we benefit from more simplification. [1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start") [2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") [3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code") Fixes: f8114f8583bb ("Revert "ALSA: usb-audio: Fix race at stopping the stream"") Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-21Revert "ALSA: usb-audio: Fix race at stopping the stream"Takashi Iwai1-8/+4
This reverts commit 16200948d8353fe29a473a394d7d26790deae0e7. The commit was intended to cover the race condition, but it introduced yet another regression for devices with the implicit feedback, leading to a kernel panic due to NULL-dereference in an irq context. As the race condition that was addressed by the commit is very rare and the regression is much worse, let's revert the commit for rc1, and fix the issue properly in a later patch. Fixes: 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") Reported-by: Ioan-Adrian Ratiu <adi@adirat.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2016-12-12ALSA: usb-audio: Eliminate noise at the start of DSD playback.Nobutaka Okabe1-1/+15
[Problem] In some USB DACs, a terrible pop noise comes to be heard at the start of DSD playback (in the following situations). - play first DSD track - change from PCM track to DSD track - change from DSD64 track to DSD128 track (and etc...) - seek DSD track - Fast-Forward/Rewind DSD track [Cause] At the start of playback, there is a little silence. The silence bit pattern "0x69" is required on DSD mode, but it is not like that. [Solution] This patch adds DSD silence pattern to the endpoint settings. Signed-off-by: Nobutaka Okabe <nob77413@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-09Merge branch 'for-linus' into for-nextTakashi Iwai1-4/+8
2016-12-06ALSA: usb-audio: more tolerant packetsizeAndreas Pape1-2/+2
since commit 57e6dae1087b ("ALSA: usb-audio: do not trust too-big wMaxPacketSize values"), the expected packetsize is always limited to nominal + 25%. It was discovered, that some devices (Android audio accessory) have a much higher jitter in used packetsizes than 25% which would result in BABBLE condition and dropping of packets. A better solution is so assume the jitter to be the nominal packetsize: -one nearly empty packet followed by a almost 150% sized one. V2: changed to assume max frequency is +50 of nominal packetsize. Signed-off-by: Andreas Pape <apape@de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-05ALSA: usb-audio: Fix race at stopping the streamTakashi Iwai1-4/+8
We've got a kernel crash report showing like: Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000 [00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] PREEMPT SMP ARM CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1 task: a3ae61c0 ti: a08c8000 task.ti: a08c8000 PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio] LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio] pc : [<7f0eb22c>] lr : [<7f0e57fc>] psr: 200e0193 sp : a08c9c98 ip : a08c9ce8 fp : a08c9ce4 r10: 0000000a r9 : 00000102 r8 : 94cb3000 r7 : 94cb3000 r6 : 94d0f000 r5 : 94d0e8e8 r4 : 94d0e000 r3 : 7f0eb21c r2 : 00000000 r1 : 94cb3000 r0 : 00000000 Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5387d Table: 31d7c04a DAC: 00000015 Process dbus-daemon (pid: 250, stack limit = 0xa08c8238) Stack: (0xa08c9c98 to 0xa08ca000) ... Backtrace: [<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio]) [<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4) [<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0) [<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148) [<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380) [<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc) [<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8) [<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8) [<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78) [<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100) [<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8) [<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c) [<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0) [<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20) Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008) Kernel panic - not syncing: Fatal exception in interrupt There is a race between retire_capture_urb() and stop_endpoints(). The latter is called at stopping the stream and it sets some endpoint fields to NULL. But its call is asynchronous, thus the pending complete callback might get called after these NULL clears, and it leads the NULL dereference like the above. The fix is to move the NULL clearance after the synchronization, i.e. wait_clear_urbs(). This is called at prepare and hw_free callbacks, so it's assured to be called before the restart of the stream or the release of the stream. Also, while we're at it, put the EP_FLAG_RUNNING flag check at the beginning of snd_complete_urb() to skip the pending complete after the stream is stopped. Fixes: b2eb950de2f0 ("ALSA: usb-audio: stop both data and sync...") Reported-by: Jiada Wang <jiada_wang@mentor.com> Reported-by: Mark Craske <Mark_Craske@mentor.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22ALSA: usb: fine-tune Tenor error compensation valueDaniel Mack1-2/+2
Users of devices affected by the Tenor feedback data error report buffer underruns, even with the +/- 0x1.0000 quirk applied. Compensating the error with 0xf000 instead seems to reliably fix that issue. See https://sourceforge.net/p/alsa/mailman/message/35230259/ Reported-and-tested-by: Norman Nolte <norman.nolte@gmx.net> Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de> Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22ALSA: usb: use TEAC UD-H01 quirk for more devicesDaniel Mack1-2/+3
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to more that are based on the Tenor 8802TL chipset. Devices built by T+A are affected too, and they apparently all use the same USB PID:PID. Extend the quirky handling for that device as well, and rename the quirks flag. Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de> Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22ALSA: usb: move udh01_fb_quirk setting to quirks.cDaniel Mack1-4/+0
That's a quirk, after all, so move it where to all the other quirks live. Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-16ALSA: usb-audio: Add sanity checks for endpoint accessesTakashi Iwai1-0/+3
Add some sanity check codes before actually accessing the endpoint via get_endpoint() in order to avoid the invalid access through a malformed USB descriptor. Mostly just checking bNumEndpoints, but in one place (snd_microii_spdif_default_get()), the validity of iface and altsetting index is checked as well. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19ALSA: USB-audio: Adjust max packet size calculation for tx_length_quirkRicard Wanderlof1-2/+10
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum sample frequency, consideration must be made for the fact that four bytes of the packet contain a length descriptor and consequently must not be counted as part of the audio data. This is corroborated by the wMaxPacketSize for this device, which is 108 bytes according for the USB playback endpoint descriptor. The frame size is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte length descriptor. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19ALSA: USB-audio: Add quirk for Zoom R16/24 playbackRicard Wanderlof1-5/+20
The Zoom R16/24 have a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. (Curiously, capture data does not contain this and requires no quirk.) The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). In order to make the quirk as unintrusive as possible, for pcm.c:prepare_playback_urb(), the isochronous packet descriptors are initially set up in the same way no matter if the quirk is enabled or not. Once it is time to actually copy the data into the outgoing packet buffer (together with the added length descriptors) the isochronous descriptors are adjusted in order take the increased payload length into account. For endpoint.c:prepare_silent_urb() it makes more sense to modify the actual function, partly because the function is less complex to start with and partly because it is not as time-critical as prepare_playback_urb() (whose bulk is run with interrupts disabled), so the (minute) additional time spent in the non-quirk case is motivated by the simplicity of having a single function for all cases. The quirk is controlled by the new tx_length_quirk member in struct snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c and endpoint.c from quirks.c in a similar manner to the txfr_quirk member in the same structs. In contrast to txfr_quirk however, the quirk is enabled directly in quirks.c:create_standard_audio_quirk() by checking the USB ID in that function. Another option would be to introduce a new QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk very plain to see in the quirk table, but it was felt that the additional code needed to implement it this way would just make the implementation more complex with no real gain. Tested with a Zoom R16, both by doing capture and playback separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together using Ardour, as well as Audacity and Qtractor together with jackd. The R24 is reportedly compatible with the R16 when used as an audio interface. Both devices share the same USB ID and have the same number of inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the patch. Regression tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Tested-by: Panu Matilainen <pmatilai@laiskiainen.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19ALSA: USB-audio: Break out creation of silent urbs from prepare_outbound_urb()Ricard Wanderlof1-19/+27
Refactoring in preparation for adding Zoom R16/24 quirk. No functional change. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-13ALSA: usb-audio: Fix max packet size calculation for USB audioRicard Wanderlof1-2/+17
Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Detailed explanation and rationale: The code before this commit had the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that if the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this means that 6 frames per packet are needed, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . (The error has not been a problem because it only results in maxsize being a bit too big which just wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long.) Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-26ALSA: usb-audio: Avoid nested autoresume callsTakashi Iwai1-4/+6
After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-09ALSA: pcm: Add snd_pcm_stop_xrun() helperTakashi Iwai1-3/+1
Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-06ALSA: usb-audio: Trigger PCM XRUN at XRUNTakashi Iwai1-3/+8
The usb-audio driver detects XRUN at its complete callback, but the actual code to trigger PCM XRUN is commented out because it caused deadlock in the past. This patch revives the PCM trigger properly. It resulted in more than just enabling snd_pcm_stop(), but it had to deduce the PCM substream with proper NULL checks and holds the stream lock around the call. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-04ALSA: usb-audio: Pass direct struct pointer instead of list_headTakashi Iwai1-5/+2
Some functions in mixer.c and endpoint.c receive list_head instead of the object itself. This is not obvious and rather error-prone. Let's pass the proper object directly instead. The functions in midi.c still receive list_head and this can't be changed since the object definition isn't exposed to the outside of midi.c, so left as is. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26ALSA: usb-audio: Fix races at disconnection and PCM closingTakashi Iwai1-3/+14
When a USB-audio device is disconnected while PCM is still running, we still see some race: the disconnect callback calls snd_usb_endpoint_free() that calls release_urbs() and then kfree() while a PCM stream would be closed at the same time and calls stop_endpoints() that leads to wait_clear_urbs(). That is, the EP object might be deallocated while a PCM stream is syncing with wait_clear_urbs() with the same EP. Basically calling multiple wait_clear_urbs() would work fine, also calling wait_clear_urbs() and release_urbs() would work, too, as wait_clear_urbs() just reads some fields in ep. The problem is the succeeding kfree() in snd_pcm_endpoint_free(). This patch moves out the EP deallocation into the later point, the destructor callback. At this stage, all PCMs must have been already closed, so it's safe to free the objects. Reported-by: Alan Stern <stern@rowland.harvard.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback dataClemens Ladisch1-1/+14
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> [mick37@gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37@gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-02-26ALSA: usb-audio: Use standard printk helpersTakashi Iwai1-13/+19
Convert with dev_err() and co from snd_printk(), etc. As there are too deep indirections (e.g. ep->chip->dev->dev), a few new local macros, usb_audio_err() & co, are introduced. Also, the device numbers in some messages are dropped, as they are shown in the prefix automatically. Signed-off-by: Takashi Iwai <tiwai@suse.de>