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commit 5d1b71226dc4d44b4b65766fa9d74492f9d4587b upstream
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae565 ("ALSA: snd-usb: add support for implicit feedback")
Fixes: ca10a7ebdff1 ("ALSA: usb-audio: FT C400 sync playback EP to capture EP")
Fixes: 5e35dc0338d8 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204")
Fixes: 17f08b0d9aaf ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II")
Fixes: 103e9625647a ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk")
Cc: stable <stable@vger.kernel.org> # 3.5
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 103e9625647ad74d201e26fb74afcd8479142a37 upstream
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sudip Mukherjee <sudipm.mukherjee@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0938ecae432e7ac8b01080c35dd81d50a1e43033 upstream.
This patch fixes audio distortion on playback for the Allen&Heath
Qu-16.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201104115717.GA19046@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 1b7ecc241a67ad6b584e071bd791a54e0cd5f097 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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This reverts commit 99703c921864a318e3e8aae74fde071b1ff35bea which is
commit f0bd62b64016508938df9babe47f65c2c727d25c upstream.
It causes a number of reported issues and a fix for it has not hit
Linus's tree yet. Revert this to resolve those problems.
Cc: Alexander Tsoy <alexander@tsoy.me>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Sasha Levin <sashal@kernel.org>
Cc: Hans de Goede <jwrdegoede@fedoraproject.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 1b34121d9f26d272b0b2334209af6b6fc82d4bf1 ]
The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 2bc16b9f3223d049b57202ee702fcb5b9b507019 upstream.
The commit a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to
separate function") introduced an error in the handling of quirks for
implicit feedback endpoints. This commit fixes this.
If a quirk successfully sets up an implicit feedback endpoint, usb-audio
no longer tries to find the implicit fb endpoint itself.
Fixes: a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to separate function")
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 377a879d9832f4ba69bd6a1fc996bb4181b1e504 ]
retire_capture_urb() may print warning messages when the given URB
doesn't align, and this may flood the system log easily.
Put the rate limit to the message for avoiding it.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <alexander.levin@microsoft.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 upstream.
Add quirk to ensure a sync endpoint is properly configured.
This patch is a fix for same symptoms on Behringer UFX1204 as patch
from Albertto Aquirre on Dec 8 2016 for Axe-Fx II.
Signed-off-by: Lassi Ylikojola <lassi.ylikojola@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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snd_pcm_ops are not supposed to change at runtime. All functions
working with snd_pcm_ops provided by <sound/pcm.h> work with
const snd_pcm_ops. So mark the non-const structs as const.
Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Make this const as it is only used in a copy operation.
Done using Coccinelle.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
changes in the core at this time while a lot of changes are found in
the driver side, unsurprisingly. Below are some highlights:
ALSA core:
- A few more hardening in ALSA timer codes
- An extension of sequencer API for advertising the card / pid
- Small fixes in compress-offload and jack layers
HD-audio:
- Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
DP-MST support
- Lots of code refactoring for sharing with ASoC SKL driver
- Regression fixes for Intel HDMI/DP
- Fixups for CX20724 codec, Lenovo AiO
USB-audio:
- Add quirk_alias option to make quirk debugging easier
- Fixes for possible Oops by malformed firmware
Firewire:
- Add support for FW-1804 in tascam driver
- Improvements / changes in card registration, multi stream handling,
etc for DICE
- Lots of code refactoring
ASoC:
- Enhancements of still ongoing topology API
- Lots of commits for Intel Skylake support including HDMI support
- A few Intel Atom driver updates for recent devices
- Lots of improvements to the Renesas drivers
- Capture support for Qualcomm drivers
- Support for TI DaVinci DRA7xxx devices
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"
* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
ALSA: mixart: silence an uninitialized variable warning
ALSA: usb-audio: Add sanity checks for endpoint accesses
ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
ALSA: hda - Limit i915 HDMI binding only for HSW and later
ALSA: hda - Fix unconditional GPIO toggle via automute
ALSA: mixart: silence unitialized variable warnings
ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
ASoC: rsnd: add simplified module explanation
ASoC: hdac_hdmi: Add broxton device ID
ASoC: Intel: Bxtn: Add Broxton PCI ID
ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
ASoC: Intel: add dmabuffer to common sst_dsp
ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
ASoC: Intel: Skylake: Fix whitepsace issues
ASoC: Intel: Skylake: Move module id defines
...
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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>
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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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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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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>
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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>
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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>
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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>
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The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The first URBs are submitted during the prepare stage. When .trigger is
called, the ALSA core saves a trigger tstamp that doesn't correspond to
the actual time when the samples are submitted. The trigger_tstamp is
now updated when the first data are submitted to avoid any time offsets.
A usb-specific trigger_tstamp_pending_update flag is used for now,
at some point the flag would need to move to the ALSA core, USB
is not the only interface where silent block transfers are programmed
as part of the prepare stage, with actual data enabled when .trigger
is called.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Denon/Marantz USB DACs need a specific vendor command to switch between PCM and
DSD mode. This patch adds a new quirk function to switch between the two modes
using the specific USB vendor command.
This patch applies to the following devices:
- Marantz SA-14S1
- Marantz HD-DAC1
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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DEBUG not defined
This (widely used) construction:
if(printk_ratelimit())
dev_dbg()
Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.
[ 533.803964] retire_playback_urb: 852 callbacks suppressed
[ 538.807930] retire_playback_urb: 852 callbacks suppressed
[ 543.811897] retire_playback_urb: 852 callbacks suppressed
[ 548.815745] retire_playback_urb: 852 callbacks suppressed
[ 553.819826] retire_playback_urb: 852 callbacks suppressed
So use dev_dbg_ratelimited() instead of this construction.
Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BugLink: http://bugs.launchpad.net/bugs/1305133
Malfunctioning or slow devices can cause a flood of dmesg SPAM.
I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour
of prior art in sound/usb/pcm.c.
WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit
+ if (printk_ratelimit() &&
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Eldad Zack <eldad@fogrefinery.com>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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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As Clemens Ladisch kindly explained:
"Please note that there are two methods to identify alternate settings:
the number, which is the value in bAlternateSetting, and the index,
which is the index in the descriptor array. There might be some wording
in the USB spec that these two values must be the same, but in reality,
[insert standard rant about firmware writers], bAlternateSetting
must be treated as a random ID value."
This patch changes the name to express the correct usage semantics.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The only call site for deactivate_endpoints() at snd_usb_hw_free().
The return value is not checked there, as it is irrelevant if it
fails on hw_free.
This patch moves the deactivation of the endpoints directly into
snd_usb_hw_free().
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch changes the way URBs are allocated and their sizes are
determined for PCM playback in the snd-usb-audio driver. Currently
the driver allocates too few URBs for endpoints that don't use
implicit sync, making underruns more likely to occur. This may be a
holdover from before I/O delays could be measured accurately; in any
case, it is no longer necessary.
The patch allocates as many URBs as possible, subject to four
limitations:
The total number of URBs for the endpoint is not allowed to
exceed MAX_URBS (which the patch increases from 8 to 12).
The total number of packets per URB is not allowed to exceed
MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is
decreased from 20 to 6.
The total duration of queued data is not allowed to exceed
MAX_QUEUE, which is decreased from 24 ms to 18 ms.
The total number of ALSA frames in the output queue is not
allowed to exceed the ALSA buffer size.
The last requirement is the hardest to implement. Currently the
number of URBs needed to fill a buffer cannot be determined in
advance, because a buffer contains a fixed number of frames whereas
the number of frames in an URB varies to match shifts in the device's
clock rate. To solve this problem, the patch changes the logic for
deciding how many packets an URB should contain. Rather than using as
many as possible without exceeding an ALSA period boundary, now the
driver uses only as many packets as needed to transfer a predetermined
number of frames. As a result, unless the device's clock has an
exceedingly variable rate, the number of URBs making up each period
(and hence each buffer) will remain constant.
The overall effect of the patch is that playback works better in
low-latency settings. The user can still specify values for
frames/period and periods/buffer that exceed the capabilities of the
hardware, of course. But for values that are within those
capabilities, the performance will be improved. For example, testing
shows that a high-speed device can handle 32 frames/period and 3
periods/buffer at 48 KHz, whereas the current driver starts to get
glitchy at 64 frames/period and 2 periods/buffer.
A side effect of these changes is that the "nrpacks" module parameter
is no longer used. The patch removes it.
Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
CC: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Daniel Mack <zonque@gmail.com>
Tested-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since the quirks all apply to implicit feedback (the source endpoint
is always a data endpoint), there's no need to set and check
a flag for it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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An implicit feedback endpoint can only be a capture source. The
consumer (sink) of the implicit feedback endpoint is therefore limited
to playback EPs.
Check if the target endpoint is a playback first and remove redundant
checks.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since implicit_fb is not changed, !implicit_fb will always
be true - it is set only after these checks.
Similarly, there's also no need to set it at the top of the function.
Change the type of implicit_fb to bool (more appropriate).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Setting the sync endpoint currently takes up about half of set_format().
Move it to a dedicated function.
No functional change.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Following general kernel style.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Code block does not compile when enabled.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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All the Roland/Edirol/BOSS USB audio devices that need implicit feedback
show this unambiguously in their descriptors, so it might be a good idea
to let the driver detect this.
This should make playback work correctly (at least with Jack) with the
following devices:
- BOSS GT-100
- BOSS JS-8 Jam Station
- Edirol M-16DX
- Roland GAIA SH-01
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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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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Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.
ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.
This patch adds support for this by adding a boolean flag to the
audio format struct.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.
The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.
To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.
The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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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Johan Hovold
Alberto Aguirre
Geoffrey D. Bennett
Hector Martin
Greg Kroah-Hartman
Alexander Tsoy
Ard van Breemen
Manuel Reinhardt
Takashi Iwai
Lassi Ylikojola
Arvind Yadav
Bhumika Goyal
Ioan-Adrian Ratiu
Daniel Girnus
Mauro Carvalho Chehab
Linus Torvalds
Shuah Khan
Ricard Wanderlof
Pierre-Louis Bossart
Jurgen Kramer
Sander Eikelenboom
Tim Gardner
Eldad Zack
Alan Stern
Clemens Ladisch
Daniel Mack
Calvin Owens