#
a39d51ff |
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13-Mar-2024 |
Johan Carlsson <johan.carlsson@teenage.engineering> |
ALSA: usb-audio: Stop parsing channels bits when all channels are found. If a usb audio device sets more bits than the amount of channels it could write outside of the map array. Signed-off-by: Johan Carlsson <johan.carlsson@teenage.engineering> Fixes: 04324ccc75f9 ("ALSA: usb-audio: add channel map support") Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
5fadc941 |
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21-Aug-2023 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Fix init call orders for UAC1 There have been reports of USB-audio driver spewing errors at the probe time on a few devices like Jabra and Logitech. The suggested fix there couldn't be applied as is, unfortunately, because it'll likely break other devices. But, the patch suggested an interesting point: looking at the current init code in stream.c, one may notice that it does initialize differently from the device setup in endpoint.c. Namely, for UAC1, we should call snd_usb_init_pitch() and snd_usb_init_sample_rate() after setting the interface, while the init sequence at parsing calls them before setting the interface blindly. This patch changes the init sequence at parsing for UAC1 (and other devices that need a similar behavior) to be aligned with the rest of the code, setting the interface at first. And, this fixes the long-standing problems on a few UAC1 devices like Jabra / Logitech, as reported, too. Reported-and-tested-by: Joakim Tjernlund <joakim.tjernlund@infinera.com> Closes: https://lore.kernel.org/r/202bbbc0f51522e8545783c4c5577d12a8e2d56d.camel@infinera.com Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20230821111857.28926-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
16f1f838 |
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04-Jan-2023 |
Takashi Iwai <tiwai@suse.de> |
Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing" This reverts commit ac5e2fb425e1121ceef2b9d1b3ffccc195d55707. The commit caused a regression on Behringer UMC404HD (and likely others). As the change was meant only as a minor optimization, it's better to revert it to address the regression. Reported-and-tested-by: Michael Ralston <michael@ralston.id.au> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/CAC2975JXkS1A5Tj9b02G_sy25ZWN-ys+tc9wmkoS=qPgKCogSg@mail.gmail.com Link: https://lore.kernel.org/r/20230104150944.24918-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
ac5e2fb4 |
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31-Aug-2022 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Drop superfluous interface setup at parsing We reset each interface that is being parsed for each stream, but this is superfluous and even can lead to spurious errors. Since the interface is set up properly at opening the endpoint for each actual stream operation, let's drop the superfluous one. Link: https://lore.kernel.org/r/20220831130021.4762-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
e53f47f6 |
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05-Sep-2022 |
Dongxiang Ke <kdx.glider@gmail.com> |
ALSA: usb-audio: Fix an out-of-bounds bug in __snd_usb_parse_audio_interface() There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and the number of it's interfaces less than 4, an out-of-bounds read bug occurs when parsing the interface descriptor for this device. Fix this by checking the number of interfaces. Signed-off-by: Dongxiang Ke <kdx.glider@gmail.com> Link: https://lore.kernel.org/r/20220906024928.10951-1-kdx.glider@gmail.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
7e1afce5 |
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31-Aug-2022 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Inform the delayed registration more properly The info message that was added in the commit a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") is actually useful to know the need for the delayed registration. However, it turned out that this doesn't catch the all cases; namely, this warned only when a PCM stream is attached onto the existing PCM instance, but it doesn't count for a newly created PCM instance. This made confusion as if there were no further delayed registration. This patch moves the check to the code path for either adding a stream or creating a PCM instance. Also, make it simpler by checking the card->registered flag instead of querying each snd_device state. Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Link: https://lore.kernel.org/r/20220831125901.4660-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c1b034a4 |
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29-Jul-2021 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Move tx_length quirk handling to quirk_flags There is another quirk for the transfer, and that's currently specific to Zoom R16/24, handled in create_standard_audio_quirk(). Let's move this also to the new quirk_flags. Link: https://lore.kernel.org/r/20210729073855.19043-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
af158a7f |
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29-Jul-2021 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Move txfr_quirk handling to quirk_flags The txfr_quirk field was meant for aligning the transfer, and it's set for certain devices in quirks-table.h. Now we can move that stuff also to the new quirk_flags gracefully, and reduce the quirks-table.h entries (that are exposed to module device table). As the quirks-table.h entries are also with the name string override, provide the corresponding entries to the usb_audio_names[] table, too. Link: https://lore.kernel.org/r/20210729073855.19043-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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c6dde8ff |
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11-Dec-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Fix control 'access overflow' errors from chmap The current channel-map control implementation in USB-audio driver may lead to an error message like "control 3:0:0:Playback Channel Map:0: access overflow" when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get callback clears the whole array no matter which count is set, and rather the false-positive detection. This patch fixes the problem by clearing only the needed array range at usb_chmap_ctl_get(). Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20201211130048.6358-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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73037c8d |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Simplify snd_usb_init_pitch() arguments A preliminary change for the later big changes. This is a minor code refactoring to drop the unnecessary arguments that can be retrieved in a different way. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-21-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
953a446b |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Simplify snd_usb_init_sample_rate() arguments A preliminary change for the later big changes. This is a minor code refactoring to drop the unnecessary arguments that can be retrieved in a different way. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-20-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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c7f90201 |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Don't set altsetting before initializing sample rate Setting the active altsetting at changing sample rate seems unrecommended. The host should deselect the altsetting at first before that, then select it again. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-18-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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54cb3190 |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Create endpoint objects at parsing phase 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>
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#
f6581c0e |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list Instead of parsing and evaluating the sync endpoint and the implicit feedback mode at each time the audio stream is opened, let's parse it once at the probe time, as the all needed information can be obtained statically from the descriptor or from the quirk. This patch extends audioformat struct to record the sync endpoint, interface and altsetting as well as the implicit feedback flag, which are filled at parsing the streams. Then, set_sync_endpoint() is much simplified just to follow the already parsed data. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-9-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
bc4e94aa |
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23-Nov-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Handle discrete rates properly in hw constraints In the current code, when the device provides the discrete sample rate tables with unusual sample rates, the driver tries to gather the whole values from the audioformat entries and create a hw-constraint rule to restrict with this single rate list. This is rather inefficient and may overlook the rates that are associated only with the certain audioformat entries. This patch improves the hw constraint setup by rewriting the existing hw_rule_rate(). The discrete sample rates (identified by rate_table and nr_rates of format entry) are checked in the existing hw_rule_rate() instead of extra rules; in the case of discrete rates, the function compares with each rate table entry and calculates the min/max values from there. For the contiguous rates, the behavior doesn't change. Along with it, snd_usb_pcm_check_knot() and snb_usb_substream rate_list field become superfluous, thus those are dropped. Tested-by: Keith Milner <kamilner@superlative.org> Tested-by: Dylan Robinson <dylan_robinson@motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
1b7ecc24 |
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10-Aug-2020 |
Hector Martin <marcan@marcan.st> |
ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109 Further investigation of the L-R swap problem on the MS2109 reveals that the problem isn't that the channels are swapped, but rather that they are swapped and also out of phase by one sample. In other words, the issue is actually that the very first frame that comes from the hardware is a half-frame containing only the right channel, and after that everything becomes offset. So introduce a new quirk field to drop the very first 2 bytes that come in after the format is configured and a capture stream starts. This puts the channels in phase and in the correct order. Cc: stable@vger.kernel.org Signed-off-by: Hector Martin <marcan@marcan.st> Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c0dbbdad |
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08-Jul-2020 |
Gustavo A. R. Silva <gustavoars@kernel.org> |
ALSA: Use fallthrough pseudo-keyword Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fallthrough#implicit-switch-case-fall-through Signed-off-by: Gustavo A. R. Silva <gustavoars@kernel.org> Link: https://lore.kernel.org/r/20200708203236.GA5112@embeddedor Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
a4aad563 |
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25-Mar-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Inform devices that need delayed registration The USB-audio driver may call snd_card_register() multiple times as its probe function is per USB interface while some USB-audio devices may provide multiple interfaces to assign different streams although they belong to the same device. This works in most cases but the registration is racy, hence it may miss the device recognition, e.g. PA doesn't see certain devices when hotplugged. The recent addition of the delayed registration quirk allows to sync the registration at the last known interface, and the previous commit added a new module option to allow the dynamic setup for that purpose. Now, this patch tries to find out and notifies for such devices that require the delayed registration. It shows a message like: Found post-registration device assignment: 1234abcd:02 If you hit this message, you can pass delayed_register module option like: snd_usb_audio.delayed_register=1234abcd:02 by just copying the last shown entry. If this works, it can be added statically in the quirk list, registration_quirks[] found at the end of sound/usb/quirks.c. Link: https://lore.kernel.org/r/20200325103322.2508-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
a01df925 |
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05-Jan-2020 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: More constifications Apply const prefix to the remaining places: the static table for the unit information, the mixer maps, the validator tables, etc. Just for minor optimization and no functional changes. Link: https://lore.kernel.org/r/20200105144823.29547-12-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
57f87706 |
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20-Aug-2019 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: More validations of descriptor units Introduce a new helper to validate each audio descriptor unit before and check the unit before actually accessing it. This should harden against the OOB access cases with malformed descriptors that have been recently frequently reported by fuzzers. The existing descriptor checks are still kept although they become superfluous after this patch. They'll be cleaned up eventually later. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
f7f53018 |
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10-Aug-2017 |
Alexander Tsoy <alexander@tsoy.me> |
ALSA: usb-audio: fix PCM device order Some cards have alternate setting with non-PCM format as the first altsetting in the interface descriptors. This confuses userspace, since alsa-lib uses device 0 by default. So lets parse interfaces in two steps: 1. Parse altsettings with PCM formats. 2. Parse altsettings with non-PCM formats. This fixes at least following cards: - Audinst HUD-mx2 - Audinst HUD-mini [ Adapted to 5.3 kernel by tiwai ] Signed-off-by: Alexander Tsoy <alexander@tsoy.me> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c1ae5e7f |
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27-Jul-2019 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Unify audioformat release code There are many open code for releasing audioformat object. Provide a unified helper and call it from the all places. Only a cleanup, no functional changes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
a6706020 |
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06-Aug-2019 |
Wenwen Wang <wenwen@cs.uga.edu> |
ALSA: usb-audio: fix a memory leak bug In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is allocated through kzalloc() before the execution goto 'found_clock'. However, this structure is not deallocated if the memory allocation for 'pd' fails, leading to a memory leak bug. To fix the above issue, free 'fp->chmap' before returning NULL. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
1a59d1b8 |
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27-May-2019 |
Thomas Gleixner <tglx@linutronix.de> |
treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 156 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>
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#
66354f18 |
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01-Apr-2019 |
Shuah Khan <shuah@kernel.org> |
media: sound/usb: Use Media Controller API to share media resources Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. Change the ALSA driver to use the Media Controller API to share media resources with DVB, and V4L2 drivers on a AU0828 media device. The Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Shuah Khan <shuah@kernel.org> Signed-off-by: Hans Verkuil <hverkuil-cisco@xs4all.nl> Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung@kernel.org>
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#
3e96d728 |
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02-Jan-2019 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Always check descriptor sizes in parser code There are a few places where we access the data without checking the actual object size from the USB audio descriptor. This may result in OOB access, as recently reported. This patch addresses these missing checks. Most of added codes are simple bLength checks in the caller side. For the input and output terminal parsers, we put the length check in the parser functions. For the input terminal, a new argument is added to distinguish between UAC1 and the rest, as they treat different objects. Reported-by: Mathias Payer <mathias.payer@nebelwelt.net> Reported-by: Hui Peng <benquike@163.com> Tested-by: Hui Peng <benquike@163.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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11175556 |
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01-Aug-2018 |
Wei Yongjun <weiyongjun1@huawei.com> |
ALSA: usb-audio: Fix invalid use of sizeof in parse_uac_endpoint_attributes() sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
a0a4959e |
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31-Jul-2018 |
Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> |
ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks Make use of UAC3 Power Domains associated to an Audio Streaming path within the PCM's logic. This means, when there is no audio being transferred (pcm is closed), the host will set the Power Domain associated to that substream to state D1. When audio is being transferred (from hw_params onwards), the Power Domain will be set to D0 state. This is the way the host lets the device know which Terminal is going to be actively used and it is for the device to manage its own internal resources on that UAC3 Power Domain. Note the resume method now sets the Power Domain to D1 state as resuming the device doesn't mean audio streaming will occur. Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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7edf3b5e |
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31-Jul-2018 |
Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> |
ALSA: usb-audio: AudioStreaming Power Domain parsing Power Domains in the UAC3 spec are mainly intended to be associated to an Input or Output Terminal so the host changes the power state of the entire capture or playback path within the topology. This patch adds support for finding Power Domains associated to an Audio Streaming Interface (bTerminalLink) and adds a reference to them in the usb audio substreams (snd_usb_substream). Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
f274baa4 |
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27-May-2018 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as this serves merely as an intermediate buffer that is copied to each URB transfer buffer. This works well in general on x86, but on some archs this may result in cache coherency issues when mmap is used. OTOH, it works also on such arch unless mmap is used. This patch is a step for mitigating the inconvenience; a new module option "use_vmalloc" is provided so that user can choose to allocate the DMA coherent buffer instead of the existing vmalloc buffer. The drawback is that it'd be the standard dma_alloc_coherent() calls and the system would require contiguous pages on non-x86 archs. Note that it's a global option and not dynamically switchable since the buffer is pre-allocated at the probe time. In theory, it's possible to be switchable, but it'd be trickier and racier. As default use_vmalloc option is set to true, so that the old behavior is kept. For allowing the coherent mmap on ARM or MIPS, pass use_vmalloc=0 option explicitly. Reported-and-tested-by: Daniel Danzberger <daniel@dd-wrt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
6cd17ea7 |
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17-May-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: stream: fix potential memory leak during uac3 interface parsing UAC3 channel map is created during interface parsing, and in some cases was not freed in failure paths. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c99f0802 |
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11-May-2018 |
Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> |
ALSA: usb-audio: Use Class Specific EP for UAC3 devices. bmAtributes offset doesn't exist in the UAC3 CS_EP descriptor. Hence, checking for pitch control as if it was UAC2 doesn't make any sense. Use the defined UAC3 offsets instead. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
17156f23 |
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03-May-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: add UAC3 BADD profiles support Recently released USB Audio Class 3.0 specification contains BADD (Basic Audio Device Definition) document which describes pre-defined UAC3 configurations. BADD support is mandatory for UAC3 devices, it should be implemented as a separate USB device configuration. As per BADD document, class-specific descriptors shall not be included in the Device’s Configuration descriptor ("inferred"), but host can guess them from BADD profile number, number of endpoints and their max packed sizes. This patch adds support of all BADD profiles from the spec Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Tested-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
eda553f4 |
|
03-May-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: stream: refactor uac3 audio interface parsing Offload snd_usb_parse_audio_interface() function which became quite long after adding UAC3 spec support. Move class-specific parts of uac3 parsing to separate function which now produce audioformat structure that is ready to be fed to snd_usb_add_audio_stream(). Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
68faa863 |
|
03-May-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: stream: refactor uac1/2 audio interface parsing Offload snd_usb_parse_audio_interface() function which became quite long after adding UAC3 spec support. Move class-specific parts of uac1/2 parsing to separate function which now produce audioformat structure that is ready to be fed to snd_usb_add_audio_stream(). This also broke Blue Microphones workaround (which relies on audioformat decoded from previous altsetting) into two parts: prepare quirk flag analyzing previous altsetting then use it with current altsetting. Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
4d47fa84 |
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03-May-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: stream: move audioformat alloc/init into separate function Offload snd_usb_parse_audio_interface() function which became quite long after adding UAC3 spec support. Move audioformat allocation and initialization into separate function, this will make easier future refactoring. Attributes left in the original func because it'll be used for UAC3 BADD profiles suport in the future There is no functional change. Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
8e0428a7 |
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24-Apr-2018 |
Michael Drake <michael.drake@codethink.co.uk> |
ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3. The channel mapping is defined by bChRelationship, not bChPurpose. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Michael Drake <michael.drake@codethink.co.uk> Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
9a2fe9b8 |
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20-Mar-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb: initial USB Audio Device Class 3.0 support Recently released USB Audio Class 3.0 specification introduces many significant changes comparing to previous versions, like - new Power Domains, support for LPM/L1 - new Cluster descriptor - changed layout of all class-specific descriptors - new High Capability descriptors - New class-specific String descriptors - new and removed units - additional sources for interrupts - removed Type II Audio Data Formats - ... and many other things (check spec) It also provides backward compatibility through multiple configurations, as well as requires mandatory support for BADD (Basic Audio Device Definition) on each ADC3.0 compliant device This patch adds initial support of UAC3 specification that is enough for Generic I/O Profile (BAOF, BAIF) device support from BADD document. Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Reviewed-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
ceb18f51 |
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18-Mar-2018 |
Ruslan Bilovol <ruslan.bilovol@gmail.com> |
ALSA: usb-audio: move audioformat quirks to quirks.c Offload USB audio interface parsing function by moving quirks to a specially designed location (quirks.c) Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
9ecb2406 |
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11-Aug-2017 |
Markus Elfring <elfring@users.sourceforge.net> |
ALSA: usb: Delete an error message for a failed memory allocation in two functions Omit an extra message for a memory allocation failure in these functions. This issue was detected by using the Coccinelle software. Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdf Signed-off-by: Markus Elfring <elfring@users.sourceforge.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c89178f5 |
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31-Mar-2016 |
Mauro Carvalho Chehab <mchehab@kernel.org> |
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources" Unfortunately, this patch caused several regressions at au0828 and snd-usb-audio, like this one: https://bugzilla.kernel.org/show_bug.cgi?id=115561 It also showed several troubles at the MC core that handles pretty poorly the memory protections and data lifetime management. So, better to revert it and fix the core before reapplying this change. This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources")' Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
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#
836b34a9 |
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30-Mar-2016 |
Vladis Dronov <vdronov@redhat.com> |
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and create_uaxx_quirk() functions allocate the audioformat object by themselves and free it upon error before returning. However, once the object is linked to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be double-freed, eventually resulting in a memory corruption. This patch fixes these failures in the error paths by unlinking the audioformat object before freeing it. Based on a patch by Takashi Iwai <tiwai@suse.de> [Note for stable backports: this patch requires the commit 902eb7fd1e4a ('ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()')] Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358 Reported-by: Ralf Spenneberg <ralf@spenneberg.net> Cc: <stable@vger.kernel.org> # see the note above Signed-off-by: Vladis Dronov <vdronov@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
aebb2b89 |
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02-Mar-2016 |
Shuah Khan <shuah@kernel.org> |
[media] sound/usb: Use Media Controller API to share media resources Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com> Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
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#
f67d71ae |
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21-Dec-2015 |
Geliang Tang <geliangtang@163.com> |
ALSA: usb-audio: use list_for_each_entry_continue_reverse For better readability, use list_for_each_entry_continue_reverse() in have_dup_chmap(). Signed-off-by: Geliang Tang <geliangtang@163.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
e0570446 |
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19-Oct-2015 |
Ricard Wanderlof <ricard.wanderlof@axis.com> |
ALSA: USB-audio: Add quirk for Zoom R16/24 playback 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>
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#
5ee20bc7 |
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06-Sep-2015 |
Johan Rastén <johan@oljud.se> |
ALSA: usb-audio: Change internal PCM order New PCMs will now be added to the end of the chip's PCM list instead of to the front. This changes the way streams are combined so that the first capture stream will now be merged with the first playback stream instead of the last. This fixes a problem with ASUS U7. Cards with one playback stream and cards without capture streams should be unaffected by this change. Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf Signed-off-by: Johan Rastén <johan@oljud.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
0ba41d91 |
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26-Feb-2014 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Use standard printk helpers 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>
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#
71373fdd |
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10-Nov-2013 |
Anssi Hannula <anssi.hannula@iki.fi> |
ALSA: usb: Fix wrong mapping of RLC and RRC channels According to USB Audio spec v2 bits 25 and 26 of bmChannelConfig are "Back Left of Center - BLC" and "Back Right of Center - BRC", respectively. They are currently assigned to ALSA channels BLC/BRC. However, the ALSA BLC/BRC are actually the rather nonsensical "bottom left center" and "bottom right center", so the channels will be assigned wrongly. The comments in the USB code are also similarly wrong, so this is not readily apparent without looking at the actual specification. Fix the channel mapping by mapping bits 25 and 26 to RLC (Rear Left Center) and RRC (Rear Right Center), respectively, instead. Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
504333df |
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04-Nov-2013 |
David Henningsson <david.henningsson@canonical.com> |
ALSA: usb - Don't trust the channel config if the channel count changed In case the channel count of the input terminal is not the same as the channel count of the streaming descriptor, the channel config of the input terminal can not be trusted. Instead fall back to a default (guessed) channel map. This was found on a Logitech USB Headset. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
e3e35f75 |
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04-Nov-2013 |
David Henningsson <david.henningsson@canonical.com> |
ALSA: usb - For class 2 devices, use channel map from altsettings The channel config from the streaming descriptor is probably a better indicator of the channel map than the input terminal. Use the input terminal's channel map as fallback only. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
0dca01c3 |
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04-Nov-2013 |
David Henningsson <david.henningsson@canonical.com> |
ALSA: usb: supply channel maps even when wChannelConfig is unspecified If wChannelconfig is given for some formats but not others, userspace might not be able to set the channel map. This is RFC because I'm not sure what the best behaviour is - to guess the channel map from the given number of channels (it's quite likely that one channel is MONO and two channels is FL FR), or just to supply UNKNOWN for all channels. But the complete lack of channel map for a format leads userspace to believe that the format is not available at all. Or am I misunderstanding how this should be used? Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
aafe77cc |
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31-Mar-2013 |
Clemens Ladisch <clemens@ladisch.de> |
ALSA: usb-audio: add support for many Roland/Yamaha devices Add quirks to detect the various vendor-specific descriptors used by Roland and Yamaha in most of their recent USB audio and MIDI devices. Together with the previous patch, this should add audio/MIDI support for the following USB devices: - Edirol motion dive .tokyo performance package - Roland MC-808 Synthesizer - Roland BK-7m Synthesizer - Roland VIMA JM-5/8 Synthesizer - Roland SP-555 Sequencer - Roland V-Synth GT Synthesizer - Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ - Edirol V-Mixer M-200i/300/380/400/480/R-1000 - BOSS GT-10B Effects Processor - Roland Fantom G6/G7/G8 Keyboard - Cakewalk Sonar V-Studio 20/100/700 Audio Interface - Roland GW-8 Keyboard - Roland AX-Synth Keyboard - Roland JUNO-Di/STAGE/Gi Keyboard - Roland VB-99 Effects Processor - Cakewalk UM-2G MIDI Interface - Roland A-500S Keyboard - Roland SD-50 Synthesizer - Roland OCTAPAD SPD-30 Controller - Roland Lucina AX-09 Synthesizer - BOSS BR-800 Digital Recorder - Roland DUO/TRI-CAPTURE (EX) Audio Interface - BOSS RC-300 Loop Station - Roland JUPITER-50/80 Keyboard - Roland R-26 Recorder - Roland SPD-SX Controller - BOSS JS-10 Audio Player - Roland TD-11/15/30 Drum Module - Roland A-49/88 Keyboard - Roland INTEGRA-7 Synthesizer - Roland R-88 Recorder Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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#
8f898e92 |
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31-Jan-2013 |
Clemens Ladisch <clemens@ladisch.de> |
ALSA: usb-audio: store protocol version in struct audioformat Instead of reading bInterfaceProtocol from the descriptor whenever it's needed, store this value in the audioformat structure. Besides simplifying some code, this will allow us to correctly handle vendor- specific devices where the descriptors are marked with other values. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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#
ebfc594c |
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24-Apr-2013 |
Daniel Mack <zonque@gmail.com> |
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually stuffed directly after the standard USB endpoint descriptor, and this is where the driver currently expects it to be. There are, however, devices in the wild that have it the other way around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes *before* the standard enpoint. Devices known to implement it that way are "Sennheiser BTD-500" and Plantronics USB headsets. When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to change sample rates, as the bitmask for the validity of this command is storen in bmAttributes of that descriptor. Fix this by searching the entire interface instead of just the extra bytes of the first endpoint, in case the latter fails. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Torstein Hegge <hegge@resisty.net> Reported-and-tested-by: Yves G <alsa-user@vivigatt.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
1539d4f8 |
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12-Apr-2013 |
Calvin Owens <jcalvinowens@gmail.com> |
ALSA: usb: Add quirk for 192KHz recording on E-Mu devices When recording at 176.2KHz or 192Khz, the device adds a 32-bit length header to the capture packets, which obviously needs to be ignored for recording to work properly. Userspace expected: L0 L1 L2 R0 R1 R2 ...but actually got: R2 L0 L1 L2 R0 R1 Also, the last byte of the length header being interpreted as L0 of the first sample caused spikes every 0.5ms, resulting in a loud 16KHz tone (about the highest 'B' on a piano) being present throughout captures. Tested at all sample rates on an E-Mu 0404USB, and tested for regressions on a generic USB headset. Signed-off-by: Calvin Owens <jcalvinowens@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
88766f04 |
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03-Apr-2013 |
Eldad Zack <eldad@fogrefinery.com> |
ALSA: usb-audio: convert list_for_each to entry variant Change occurances of list_for_each into list_for_each_entry where applicable. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
2fcdb06d |
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17-Mar-2013 |
Daniel Mack <zonque@gmail.com> |
ALSA: snd-usb: handle the bmFormats field as unsigned int This field may use up to 32 bits, so it should be handled as unsigned int. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
04324ccc |
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26-Nov-2012 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: add channel map support Add the support for channel maps of the PCM streams on USB audio devices. The channel map information is already found in ChannelConfig descriptor entries, which haven't been referred until now. Each chmap entry is added to audioformat list entry and copied to TLV dynamically instead of creating a whole chmap array. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
978520b7 |
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12-Oct-2012 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Fix races at disconnection Close some races at disconnection of a USB audio device by adding the chip->shutdown_mutex and chip->shutdown check at appropriate places. The spots to put bandaids are: - PCM prepare, hw_params and hw_free - where the usb device is accessed for communication or get speed, in mixer.c and others; the device speed is now cached in subs->speed instead of accessing to chip->dev The accesses in PCM open and close don't need the mutex protection because these are already handled in the core PCM disconnection code. The autosuspend/autoresume codes are still uncovered by this patch because of possible mutex deadlocks. They'll be covered by the upcoming change to rwsem. Also the mixer codes are untouched, too. These will be fixed in another patch, too. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
8260ef07 |
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08-Jun-2012 |
Takashi Iwai <tiwai@suse.de> |
ALSA: usb-audio: Fix substream assignments In 3.5 kernel, the endpoint is assigned dynamically for the substreams, but the PCM assignment still checks the presence of the endpoint pointer. This ended up in duplicated PCM substream creations at probing time, resulting in kernel warnings like: WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6() Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2 Call Trace: [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48 [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b [<ffffffff811292f0>] proc_register+0x169/0x1a6 [<ffffffff8112962e>] create_proc_entry+0x74/0x8c [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd] [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm] [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio] [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio] [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio] [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio] [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio] ..... This patch fixes the regression by checking the fixed endpoint number for each substream instead of the endpoint pointer. Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
edcd3633 |
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12-Apr-2012 |
Daniel Mack <zonque@gmail.com> |
ALSA: snd-usb: switch over to new endpoint streaming logic With the previous commit that added the new streaming model, all endpoint and streaming related code is now in endpoint.c, and pcm.c only acts as a wrapper for handling the packet's payload. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
c731bc96 |
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13-Sep-2011 |
Daniel Mack <zonque@gmail.com> |
ALSA: snd-usb: move code from urb.c to endpoint.c No code altered at this point, simply preparing for upcoming refactorizations. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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#
e8e8babf |
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12-Sep-2011 |
Daniel Mack <zonque@gmail.com> |
ALSA: snd-usb: re-order code Move code from endpoint.c into a new file called stream.c and rename functions so that their names actually reflect what they're doing. This way, endpoint.c will be available to functions that hold all the endpoint logic. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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