Commit graph

123 commits

Author SHA1 Message Date
Thomas Gleixner
1a59d1b8e0 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>
2019-05-30 11:26:35 -07:00
Shuah Khan
66354f18fe 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>
2019-04-22 11:21:06 -04:00
Manuel Reinhardt
a634090a0f ALSA: usb-audio: Add quirk for MOTU MicroBook II
Add an entry to the quirks-table to for usb-audio to recognize the
Microbook II (although it only exposes vendor interfaces). A simple boot
quirk is also implemented to set up the sample rate and  make sure that
no audio urbs are sent before the device is ready.

This patch only provides audio playback and capture at 96kHz sample
rate. Notice the following shortcomings:

- The sample rate is currently hardcoded to 96k although the device also
  supports 48k and 44.1k.

- The various mixer controls of the MicroBook are not made available.

- The keep-iface control should be on by default because the device
  shuts down whenever the altsetting is reset which is usually unwanted.
  (I don't know the best way to do this)

- The communication format used by the MicroBook for sample rate setting
  and also other setup has been reverse engineered by looking at the
  usbmon output while running the windows driver through virtualbox. In
  this patch the first byte of every message is set to \0 while in the
  observed communications the first byte acts as a "message-counter"
  increasing its value with every message sent. Leaving it at \0 does
  not seem to affect the device.

Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2019-02-28 22:23:11 +01:00
Manuel Reinhardt
2bc16b9f32 ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
The commit a60945fd08 ("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: a60945fd08 ("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>
2019-02-07 20:04:43 +01:00
Gustavo A. R. Silva
d5e77fca87 ALSA: usb: Mark expected switch fall-through
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.

Addresses-Coverity-ID: 115084 ("Missing break in switch")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 20:32:06 +02:00
Jorge Sanjuan
a0a4959eb4 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>
2018-07-31 15:01:45 +02:00
Jorge Sanjuan
3f59aa11c6 ALSA: usb-audio: Add UAC3 Power Domains to suspend/resume
Set the UAC3 Power Domain state for an Audio Streaming interface
to D2 state before suspending the device (usb_driver callback).
This lets the device know there is no intention to use any of the
Units in the Audio Function and that the host is not going to
even listen for wake-up events (interrupts) on the units.

When the usb_driver gets resumed, the state D0 (fully powered) will
be set. This ties up the UAC3 Power Domains to the runtime PM.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-31 15:01:36 +02:00
Takashi Iwai
fa84cf094e ALSA: pcm: Nuke snd_pcm_lib_mmap_vmalloc()
snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL.  As the situation has never
changed over decades, let's rip it off.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-18 08:24:29 +02:00
Kees Cook
6da2ec5605 treewide: kmalloc() -> kmalloc_array()
The kmalloc() function has a 2-factor argument form, kmalloc_array(). This
patch replaces cases of:

        kmalloc(a * b, gfp)

with:
        kmalloc_array(a * b, gfp)

as well as handling cases of:

        kmalloc(a * b * c, gfp)

with:

        kmalloc(array3_size(a, b, c), gfp)

as it's slightly less ugly than:

        kmalloc_array(array_size(a, b), c, gfp)

This does, however, attempt to ignore constant size factors like:

        kmalloc(4 * 1024, gfp)

though any constants defined via macros get caught up in the conversion.

Any factors with a sizeof() of "unsigned char", "char", and "u8" were
dropped, since they're redundant.

The tools/ directory was manually excluded, since it has its own
implementation of kmalloc().

The Coccinelle script used for this was:

// Fix redundant parens around sizeof().
@@
type TYPE;
expression THING, E;
@@

(
  kmalloc(
-	(sizeof(TYPE)) * E
+	sizeof(TYPE) * E
  , ...)
|
  kmalloc(
-	(sizeof(THING)) * E
+	sizeof(THING) * E
  , ...)
)

// Drop single-byte sizes and redundant parens.
@@
expression COUNT;
typedef u8;
typedef __u8;
@@

(
  kmalloc(
-	sizeof(u8) * (COUNT)
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(__u8) * (COUNT)
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(char) * (COUNT)
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(unsigned char) * (COUNT)
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(u8) * COUNT
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(__u8) * COUNT
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(char) * COUNT
+	COUNT
  , ...)
|
  kmalloc(
-	sizeof(unsigned char) * COUNT
+	COUNT
  , ...)
)

// 2-factor product with sizeof(type/expression) and identifier or constant.
@@
type TYPE;
expression THING;
identifier COUNT_ID;
constant COUNT_CONST;
@@

(
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * (COUNT_ID)
+	COUNT_ID, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * COUNT_ID
+	COUNT_ID, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * (COUNT_CONST)
+	COUNT_CONST, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * COUNT_CONST
+	COUNT_CONST, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * (COUNT_ID)
+	COUNT_ID, sizeof(THING)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * COUNT_ID
+	COUNT_ID, sizeof(THING)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * (COUNT_CONST)
+	COUNT_CONST, sizeof(THING)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * COUNT_CONST
+	COUNT_CONST, sizeof(THING)
  , ...)
)

// 2-factor product, only identifiers.
@@
identifier SIZE, COUNT;
@@

- kmalloc
+ kmalloc_array
  (
-	SIZE * COUNT
+	COUNT, SIZE
  , ...)

// 3-factor product with 1 sizeof(type) or sizeof(expression), with
// redundant parens removed.
@@
expression THING;
identifier STRIDE, COUNT;
type TYPE;
@@

(
  kmalloc(
-	sizeof(TYPE) * (COUNT) * (STRIDE)
+	array3_size(COUNT, STRIDE, sizeof(TYPE))
  , ...)
|
  kmalloc(
-	sizeof(TYPE) * (COUNT) * STRIDE
+	array3_size(COUNT, STRIDE, sizeof(TYPE))
  , ...)
|
  kmalloc(
-	sizeof(TYPE) * COUNT * (STRIDE)
+	array3_size(COUNT, STRIDE, sizeof(TYPE))
  , ...)
|
  kmalloc(
-	sizeof(TYPE) * COUNT * STRIDE
+	array3_size(COUNT, STRIDE, sizeof(TYPE))
  , ...)
|
  kmalloc(
-	sizeof(THING) * (COUNT) * (STRIDE)
+	array3_size(COUNT, STRIDE, sizeof(THING))
  , ...)
|
  kmalloc(
-	sizeof(THING) * (COUNT) * STRIDE
+	array3_size(COUNT, STRIDE, sizeof(THING))
  , ...)
|
  kmalloc(
-	sizeof(THING) * COUNT * (STRIDE)
+	array3_size(COUNT, STRIDE, sizeof(THING))
  , ...)
|
  kmalloc(
-	sizeof(THING) * COUNT * STRIDE
+	array3_size(COUNT, STRIDE, sizeof(THING))
  , ...)
)

// 3-factor product with 2 sizeof(variable), with redundant parens removed.
@@
expression THING1, THING2;
identifier COUNT;
type TYPE1, TYPE2;
@@

(
  kmalloc(
-	sizeof(TYPE1) * sizeof(TYPE2) * COUNT
+	array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2))
  , ...)
|
  kmalloc(
-	sizeof(TYPE1) * sizeof(THING2) * (COUNT)
+	array3_size(COUNT, sizeof(TYPE1), sizeof(TYPE2))
  , ...)
|
  kmalloc(
-	sizeof(THING1) * sizeof(THING2) * COUNT
+	array3_size(COUNT, sizeof(THING1), sizeof(THING2))
  , ...)
|
  kmalloc(
-	sizeof(THING1) * sizeof(THING2) * (COUNT)
+	array3_size(COUNT, sizeof(THING1), sizeof(THING2))
  , ...)
|
  kmalloc(
-	sizeof(TYPE1) * sizeof(THING2) * COUNT
+	array3_size(COUNT, sizeof(TYPE1), sizeof(THING2))
  , ...)
|
  kmalloc(
-	sizeof(TYPE1) * sizeof(THING2) * (COUNT)
+	array3_size(COUNT, sizeof(TYPE1), sizeof(THING2))
  , ...)
)

// 3-factor product, only identifiers, with redundant parens removed.
@@
identifier STRIDE, SIZE, COUNT;
@@

(
  kmalloc(
-	(COUNT) * STRIDE * SIZE
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	COUNT * (STRIDE) * SIZE
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	COUNT * STRIDE * (SIZE)
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	(COUNT) * (STRIDE) * SIZE
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	COUNT * (STRIDE) * (SIZE)
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	(COUNT) * STRIDE * (SIZE)
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	(COUNT) * (STRIDE) * (SIZE)
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
|
  kmalloc(
-	COUNT * STRIDE * SIZE
+	array3_size(COUNT, STRIDE, SIZE)
  , ...)
)

// Any remaining multi-factor products, first at least 3-factor products,
// when they're not all constants...
@@
expression E1, E2, E3;
constant C1, C2, C3;
@@

(
  kmalloc(C1 * C2 * C3, ...)
|
  kmalloc(
-	(E1) * E2 * E3
+	array3_size(E1, E2, E3)
  , ...)
|
  kmalloc(
-	(E1) * (E2) * E3
+	array3_size(E1, E2, E3)
  , ...)
|
  kmalloc(
-	(E1) * (E2) * (E3)
+	array3_size(E1, E2, E3)
  , ...)
|
  kmalloc(
-	E1 * E2 * E3
+	array3_size(E1, E2, E3)
  , ...)
)

// And then all remaining 2 factors products when they're not all constants,
// keeping sizeof() as the second factor argument.
@@
expression THING, E1, E2;
type TYPE;
constant C1, C2, C3;
@@

(
  kmalloc(sizeof(THING) * C2, ...)
|
  kmalloc(sizeof(TYPE) * C2, ...)
|
  kmalloc(C1 * C2 * C3, ...)
|
  kmalloc(C1 * C2, ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * (E2)
+	E2, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(TYPE) * E2
+	E2, sizeof(TYPE)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * (E2)
+	E2, sizeof(THING)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	sizeof(THING) * E2
+	E2, sizeof(THING)
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	(E1) * E2
+	E1, E2
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	(E1) * (E2)
+	E1, E2
  , ...)
|
- kmalloc
+ kmalloc_array
  (
-	E1 * E2
+	E1, E2
  , ...)
)

Signed-off-by: Kees Cook <keescook@chromium.org>
2018-06-12 16:19:22 -07:00
Takashi Iwai
f274baa49b 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>
2018-05-29 10:01:54 +02:00
Takashi Iwai
f25ecf8f98 ALSA: usb-audio: Follow standard coding style
Avoid if ((err = ...) style and expand to multiple lines instead.
No change in the end result, but just the beautification.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28 14:52:40 +02:00
Takashi Iwai
e92be8146c ALSA: usb-audio: Move autoresume call at the end of open
... so that we can avoid the extra goto lines.
Also beautify the code to follow the standard codex.

No functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28 14:50:50 +02:00
Takashi Iwai
6fddc79787 ALSA: usb-audio: Simplify PCM open/close callbacks
The stream direction in open and close callbacks can be retrieved from
substream->direction, hence we don't have to stick with the unique PCM
ops hard-coded for each direction.  Rewrite the common open/close
callback functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-28 11:53:04 +02:00
Takashi Iwai
377a879d98 ALSA: usb-audio: Apply rate limit to warning messages in URB complete callback
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>
2018-05-16 20:07:18 +02:00
Takashi Iwai
8a463225b1 ALSA: usb-audio: Add keep_iface flag
Introduce a new flag to struct snd_usb_audio for allowing the device
to skip usb_set_interface() calls at changing or closing the stream.
As of this patch, the flag is nowhere set, so it's just a place
holder.  The dynamic switching will be added in the following patch.

A background information for this change:

Dell WD15 dock with Realtek chip gives a very long pause at each time
the driver changes the altset, which eventually happens at every PCM
stream open/close and parameter change.  As the long pause happens in
each usb_set_interface() call, there is nothing we can do as long as
it's called.  The workaround is to reduce calling it as much as
possible, and this flag indicates that behavior.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-02 16:02:33 +02:00
Takashi Iwai
b099b9693d ALSA: usb-audio: Avoid superfluous usb_set_interface() calls
This is a preliminary change for the upcoming quirk implementation.

Currently USB-audio driver tries to call usb_set_interface() whenever
the format change with interface/altset modification happens.  In this
patch, the check is replaced with the comparison of cur_altsetting and
the targeted altsetting pointer, so that the driver may skip the
unnecessary function calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-05-02 16:02:33 +02:00
Alberto Aguirre
91a8561d0e ALSA: usb-audio: add implicit fb quirk for Axe-Fx III
The Axe-Fx III implicit feedback end point and the data sink 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>
2018-04-19 11:49:29 +02:00
Alberto Aguirre
103e962564 ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-04-19 11:49:22 +02:00
Lassi Ylikojola
5e35dc0338 ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204
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>
2018-02-12 08:21:28 +01:00
Arvind Yadav
31cb1fb41d ALSA: usb: constify snd_pcm_ops structures
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>
2017-08-19 11:02:27 +02:00
Bhumika Goyal
aaffbf7824 ALSA: usb: make snd_pcm_hardware const
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>
2017-08-17 12:44:23 +02:00
Ioan-Adrian Ratiu
1d0f953086 ALSA: usb-audio: Fix irq/process data synchronization
Commit 16200948d8 ("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: f8114f8583 ("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>
2017-01-05 07:35:00 +01:00
Alberto Aguirre
17f08b0d9a ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
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>
2016-12-09 11:19:31 +01:00
Daniel Girnus
1e2e3fe480 ALSA: usb-audio: avoid setting of sample rate multiple times on bus
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>
2016-12-06 13:55:15 +01:00
Mauro Carvalho Chehab
c89178f57a [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 aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'

Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
2016-03-31 15:02:33 -03:00
Linus Torvalds
021f163d69 sound updates for 4.6-rc1
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
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Merge tag 'sound-4.6-rc1' of 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
  ...
2016-03-18 10:05:46 -07:00
Takashi Iwai
447d6275f0 ALSA: usb-audio: Add sanity checks for endpoint accesses
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>
2016-03-16 12:45:32 +01:00
Shuah Khan
aebb2b89bf [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>
2016-03-03 15:01:13 -03:00
Ricard Wanderlof
e057044677 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>
2015-10-19 12:38:09 +02:00
Ricard Wanderlof
b97a936910 ALSA: USB-audio: Add offset parameter to copy_to_urb()
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>
2015-10-19 12:38:08 +02:00
Ricard Wanderlof
4c4e4391b8 ALSA: USB-audio: Also move out hwptr_done wrap from prepare_playback_urb()
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>
2015-10-19 12:38:07 +02:00
Ricard Wanderlof
07a40c2fc6 ALSA: USB-audio: Break out copying to urb from prepare_playback_urb()
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>
2015-10-19 12:38:06 +02:00
Takashi Iwai
47ab154593 ALSA: usb-audio: Avoid nested autoresume calls
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: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-26 15:38:25 +02:00
Pierre-Louis Bossart
395ae54bd8 ALSA: usb: handle descriptor with SYNC_NONE illegal value
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>
2015-08-16 08:48:47 +02:00
Pierre-Louis Bossart
630184477e ALSA: usb: fix corrupted pointers due to interface setting change
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>
2015-08-16 08:48:35 +02:00
Pierre-Louis Bossart
ea33d359c4 ALSA: usb: update trigger timestamp on first non-zero URB submitted
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>
2015-02-09 16:02:43 +01:00
Jurgen Kramer
6874daad4b ALSA: usb-audio: Add mode select quirk for Denon/Marantz DACs
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>
2014-11-28 18:02:35 +01:00
Sander Eikelenboom
b7a7723513 ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while 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>
2014-05-02 18:10:59 +02:00
Tim Gardner
a5065eb6da ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
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>
2014-04-09 21:07:38 +02:00
Takashi Iwai
0ba41d917e 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>
2014-02-26 16:45:34 +01:00
Eldad Zack
df23a2466a ALSA: usb-audio: rename alt_idx to altsetting
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>
2013-10-07 11:22:03 +02:00
Eldad Zack
06613f547a ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error
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>
2013-10-07 11:00:23 +02:00
Eldad Zack
26de5d0a8d ALSA: usb-audio: remove deactivate_endpoints()
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>
2013-10-07 10:52:13 +02:00
Alan Stern
976b6c064a ALSA: improve buffer size computations for USB PCM audio
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>
2013-09-26 10:25:31 +02:00
Eldad Zack
88abb8eff4 ALSA: usb-audio: remove implicit_fb from quirk
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>
2013-08-06 10:52:14 +02:00
Eldad Zack
914273c714 ALSA: usb-audio: remove is_playback from implicit feedback quirks
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>
2013-08-06 10:51:48 +02:00
Eldad Zack
95fec88332 ALSA: usb-audio: do not initialize and check implicit_fb
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>
2013-08-06 10:51:11 +02:00
Eldad Zack
f34d065013 ALSA: usb-audio: reverse condition logic in set_sync_endpoint
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>
2013-08-06 10:50:15 +02:00
Eldad Zack
a60945fd08 ALSA: usb-audio: move implicit fb quirks to separate function
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>
2013-08-06 10:49:21 +02:00
Eldad Zack
71bb64c56d ALSA: usb-audio: separate sync endpoint setting from set_format
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>
2013-08-06 10:48:34 +02:00