Commit graph

547686 commits

Author SHA1 Message Date
Takashi Iwai
b9a94a9c78 ALSA: hda - convert to hda_device_id
Finally we have a proper infrastructure to generate the modaliases
automatically, let's move to hda_device_id from the legacy
hda_codec_preset that contains basically the same information.

The patch function hook is stored in driver_data field, which is long,
and we need an explicit cast.  Other than that, the conversion is
mostly straightforward.  Each entry is even simplified using a macro,
and the lengthy (and error-prone) manual modaliases got removed.

As a result, we achieved a quite good diet:
 14 files changed, 407 insertions(+), 595 deletions(-)

Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-20 10:15:20 +02:00
Subhransu S. Prusty
78abb2afaf ALSA: hda - Add hdaudio bus modalias support
This patch just adds modalias sysfs entry to each hdaudio bus entry.

[rewritten to call the common helper function by tiwai]

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-20 10:15:09 +02:00
Takashi Iwai
4f9e0c38c5 ALSA: hda - Add a common helper to give the codec modalias string
This patch provide a new common helper function,
snd_hdac_codec_modalias(), to give the codec modalias name string.
This function will be used by multiple places in the later patches.

Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-20 10:14:59 +02:00
Subhransu S. Prusty
da23ac1e40 ALSA: hda - Add hduadio support to DEVTABLE
For generating modalias entries automatically, move the definition of
struct hda_device_id to linux/mod_devicetable.h and add the handling
of this record in file2alias helper.  The new modalias is represented
with combination of vendor id, device id, and api version as
"hdaudio:vNrNaN".

This patch itself doesn't convert the existing modaliases.  Since they
were added manually, this patch won't give any regression by itself at
this point.

[Modified the modalias format to adapt the api_version field, and drop
 invalid ANY_ID definition by tiwai]

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-20 10:14:42 +02:00
Takashi Iwai
93ed8560e9 ALSA: hda - Add api_version to hda_device_id struct
For distinguishing the difference between HDA legacy and ext codec
driver entries, we need to expose the value corresponding to type
field.  This patch adds a new field, api_version, to hda_device_id
struct, so that this information is embedded in modalias string.

Although the information is basically redundant (struct hdac_device
already has type field), the helper that extracts from
MODULE_DEVICE_TABLE() won't take it account except for the exported
table entries themselves.  So we need to put the same information in
the table, too.

Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-20 10:14:17 +02:00
Takashi Iwai
7fbe824a0f ALSA: hda - Update mixer name for the lower codec address
In most cases, we prefer the onboard codec as the primary device, thus
it's better to set it as the mixer name.  Currently, however, the
mixer name is updated per the device instantiation order, and user
gets often HDMI/DP or other seen as a mixer chip name.  Also, if a
codec name is renamed by the driver, the old chip name might be left
still as the mixer name.

This patch addresses these issues by remembering the chip address that
was referred as the mixer name.  When a codec with the same or lower
address gives its name, renew the mixer name accordingly, as it's
either the update of the codec name or we get likely the more
appropriate chip as the reference.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 14:10:25 +02:00
Takashi Iwai
ded255be22 ALSA: hda - consolidate chip rename functions
A few multiple codec drivers do renaming the chip_name string but all
these are open-coded and some of them have even no error check.  Let's
make common helpers to do it properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 14:05:28 +02:00
Takashi Iwai
3e19fec33a ALSA: hda - Enable widget power saving for Cirrus codecs
Cirrus codecs have also fine power controls on each widget, thus it
gets benefit from the recent widget power-saving feature.  As we
haven't seen any obvious regressions with tests on some MacBooks,
let's try to enable it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 11:19:39 +02:00
Mikko Rapeli
ffc287c891 ALSA: hdspm: use __u8, __u32 and __u64 from linux/types.h instead of stdint.h
Kernel headers should use linux/types.h based definitions.

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 10:03:11 +02:00
Mikko Rapeli
a82d24f83d ALSA: emu10k1: added EMU10K1 version of DECLARE_BITMAP macro for UAPI
Fixes userspace compilation error:

error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’
  DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */

DECLARE_BITMAP macro is not meant for userspace headers and thus
added here as private copy for emu10k.h.

Fix was suggested by Arnd Bergmann <arnd@arndb.de> in message
<2168807.4Yxh5gl11Q@wuerfel> and Takashi Iwai <tiwai@suse.de>
in message <s5h1thx88tk.wl-tiwai@suse.de> on lkml.

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 10:02:26 +02:00
Dan Carpenter
5a1f8c4225 ALSA: oss: underflow in snd_mixer_oss_proc_write()
We cap the upper bound of "idx" but not the negative side.  Let's make
it unsigned to fix this.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-15 10:00:29 +02:00
Ricard Wanderlof
ab30965d9b ALSA: usb-audio: Fix max packet size calculation for USB audio
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.

We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.

Detailed explanation and rationale:

The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:

	maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
			>> (16 - ep->datainterval);

Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.

The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.

In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.

The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.

Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).

This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.

The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.

For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.

Rephrasing the maxsize expression to:

	maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
			 (frame_bits >> 3);

for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.

We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):

Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56

This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .

(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)

Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-13 11:40:44 +02:00
Takashi Iwai
3c69ea4440 Merge branch 'for-linus' into for-next 2015-10-13 11:37:06 +02:00
David Henningsson
e8d65a8d98 ALSA: hda - Fix inverted internal mic on Lenovo G50-80
Add the appropriate quirk to indicate the Lenovo G50-80 has a stereo
mic input where one channel has reverse polarity.

Alsa-info available at:
https://launchpadlibrarian.net/220846272/AlsaInfo.txt

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1504778
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-13 11:34:33 +02:00
Vinod Koul
42f2bb1c49 ALSA: hdac: Explicitly add io.h
Compiling the hdac extended core on arm fails with below error:

  sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel':
>> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of
>> function
+'writel' [-Werror=implicit-function-declaration]
     writel(value, addr);
     ^
   sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl':
>> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of
>> function
+'readl' [-Werror=implicit-function-declaration]
     return readl(addr);

This is fixed by explicitly including io.h

Fixes: 99463b3a39 - ('ALSA: hda: provide default bus io ops extended hdac')
Reported-by: kbuild test robot <lkp@intel.com>
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-13 11:33:45 +02:00
Takashi Sakamoto
53b3ffee78 ALSA: firewire-tascam: change device probing processing
Currently, this driver picks up model name with be32_to_cpu() macro
to align characters. This is wrong operation because the result is
different depending on CPU endiannness.

Additionally, vendor released several versions of firmware for this
series. It's not better to assign model-dependent information to
device entry according to the version field.

This commit fixes these bugs. The name of model is picked up correctly
and used to identify model-dependent information.

Cc: Stefan Richter <stefanr@s5r6.in-berlin.de>
Fixes: c0949b2785 ('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-12 14:17:02 +02:00
Takashi Sakamoto
e65e2cb99e ALSA: firewire-tascam: Turn on/off FireWire LED
TASCAM FireWire series has some LEDs on its surface. These LEDs can be
turned on/off by receiving asynchronous transactions to a certain
address. One of the LEDs is labels as 'FireWire'. It's better to light it
up when this driver starts to work. Besides, the LED for 'FireWire' is
turned off at bus reset.

This commit implements this idea.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-12 14:16:19 +02:00
Takashi Sakamoto
0db18e7eec ALSA: firewire-tascam: add support for MIDI functionality
In former commits, this driver got functionalities to transfer/receive
MIDI messages to/from TASCAM FireWire series.

This commit adds some ALSA MIDI ports to enable userspace applications
to use the functionalities.

I note that this commit doesn't support virtual MIDI ports which console
models support. A physical controls can be assigned to a certain MIDI
ports including physical and virtual. But the way is not clear.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-12 14:16:18 +02:00
Takashi Sakamoto
3beab0f844 ALSA: firewire-tascam: add support for outgoing MIDI messages by asynchronous transaction
TASCAM FireWire series use asynchronous transaction to receive MIDI
messages. The transaction should be sent to a certain address.

This commit supports the outgoing MIDI messages. The messages in the
transaction includes some quirks:
 * One MIDI message is transferred in one quadlet transaction, except for
   system exclusives.
 * MIDI running status is not allowed, thus transactions always include
   status byte.
 * The basic data format is the same as transferring MIDI messages
   supported in previous commit.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-12 14:16:18 +02:00
Takashi Sakamoto
107cc0129a ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction
TASCAM FireWire series use asynchronous transaction to transfer MIDI
messages. The transaction is sent to a registered address.

This commit supports the incoming MIDI messages. The messages in the
transaction include some quirks:
 * Two quadlets are used for one MIDI message and one timestamp.
 * Usually, the first byte of the first quadlet includes MIDI port and MSB
   4 bit of MIDI status. For system exclusive message, the first byte
   includes MIDI port and 0x04, or 0x07 in the end of the message.
 * The rest of the first quadlet includes MIDI bytes up to 3.
 * Several set of MIDI messages and timestamp can be transferred in one
   block transaction, up to 8 sets.

I note that TASCAM FireWire series ignores ID bytes of system exclusive
message. When receiving system exclusive messages with ID bytes on physical
MIDI bus, the series transfers the messages without ID bytes on IEEE 1394
bus, and vice versa.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-12 14:16:17 +02:00
Takashi Sakamoto
e8bd577ae6 ALSA: firewire-digi00x: add support for MIDI ports for physical controls
In former commits, asynchronous transactions are supported for physical
controls. This commit adds a pair of MIDI ports for them.

This driver already adds diferrent number of ALSA MIDI ports for physical
MIDI ports, and the number of in/out ports are different. As seeing as
'amidi' program in alsa-utils package, a pair of in/out MIDI ports is
expected with the same name. Therefore, this commit adds a pair of new
ports to the first.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:26:21 +02:00
Takashi Sakamoto
b47f525f76 ALSA: firewire-digi00x: add support of asynchronous transaction for outgoing MIDI messages to physical controls
In previous commit, asynchronous transaction for incoming MIDI messages
from physical controls is supported. The physical controls may be
controlled by receiving MIDI messages at a certain address.

This commit supports asynchronous transaction for this purpose.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:26:14 +02:00
Takashi Sakamoto
3646a54acd ALSA: firewire-digi00x: add support of asynchronous transaction for incoming MIDI messages from physical controls
Digi 00x series has two types of model; rack and console. The console
models have physical controls. The model can transmit control messages.
These control messages are transferred by asynchronous transactions to
registered address.

This commit supports the asynchronous transaction.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:26:09 +02:00
Takashi Sakamoto
9fbfd38b20 ALSA: firewire-digi00x: add support for MIDI ports corresponding to isochronous packet streaming
This commit adds MIDI functionality to capture/playback MIDI messages
from/to physical MIDI ports. These messages are transferred in isochronous
packets.

When no substreams request AMDTP streams to run, this driver starts the
streams at current sampling rate. When other substreams start at different
sampling rate, the streams are stopped temporarily, then start again at
requested sampling rate. This operation can generate missing MIDI bytes,
thus it's preferable to start PCM substreams at favorite sampling rate in
advance.

Digi 002/003 console also has a set of MIDI port for physical controls.
These ports are added in later commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:26:04 +02:00
Takashi Sakamoto
9dc5d31cdc ALSA: firewire-digi00x: handle MIDI messages in isochronous packets
In Digi 002/003 protocol, MIDI messages are transferred in the last data
channel of data blocks. Although this data channel has a label of 0x80,
it's not fully MIDI conformant data channel especially because the Counter
field always zero independently of included MIDI bytes. The 4th byte of
the data channel in LSB tells the number of included MIDI bytes. This byte
also includes the number of MIDI port. Therefore, the data format in this
data channel is:
 * 1st: 0x80 as label
 * 2nd: MIDI bytes
 * 3rd: 0 or MIDI bytes
 * 4th: the number of MIDI byte and the number of MIDI port

This commit adds support of MIDI messages in data block processing layer.

Like AM824 data format, this data channel has a capability to transfer
more MIDI messages than the capability of phisical MIDI bus. Therefore, a
throttle for data rate is  required to prevent devices' internal buffer to
overflow.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:25:57 +02:00
Takashi Sakamoto
17385a386c ALSA: firewire-digi00x: use in-kernel representation for the type of 8 bits
Original code for 'DoubleOhThree' encoding was written with '__u8' type,
while the type is usually used to export something to userspace.

This commit replaces the type with 'u8'.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:25:46 +02:00
Keith A. Milner
ac77423609 ALSA: usb-audio: Allow any MIDI endpoint to drive use of interrupt transfer on newer Roland devices
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.

This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.

It would benefit from some regresison testing with other devices if
possible.

Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:18:59 +02:00
Takashi Sakamoto
2a7e1713cd ALSA: firewire-lib: continue packet processing at detecting wrong CIP headers
In firewire-lib, isochronous packet streaming is stopped when detecting
wrong value for FMT field of CIP headers. Although this is appropriate
to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization
use invalid value to FMT field of CIP headers in the beginning of
streaming.

$ journalctl
  snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff

I got this log with M-Audio FireWire 1814. In this line, the value of FMT
field is 0x00, while it should be 0x10 in usual AMDTP.

Except for the beginning, these devices continue to transfer packets with
valid value for FMT field, except for the beginning. Therefore, in this
case, firewire-lib should continue to process packets. The former
implementation of firewire-lib performs it.

This commit loosens the handling of wrong value, to continue packet
processing in the case.

Fixes: 414ba022a5 ('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:14:01 +02:00
Julia Lawall
6b9866c893 ALSA: bebob: constify various snd_bebob structures
The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec,
snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are
initialized.  Make them all const.

Done with the help of Coccinelle.

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-11 18:12:37 +02:00
Takashi Sakamoto
bde3e2880f ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error
Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.

This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:06 +02:00
Takashi Sakamoto
ea848b7b62 ALSA: firewire-lib: add throttle for MIDI data rate
Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.

This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:06 +02:00
Takashi Sakamoto
e8a40d9bcb ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages
Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.

This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:05 +02:00
Takashi Sakamoto
d3ef9cb93a ALSA: firewire-lib: add a restriction for a transaction at once
Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.

This commit serialize request/response transactions, by adding one
boolean member to represent idling state.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:05 +02:00
Takashi Sakamoto
585d7cba5e ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages
Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.

This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.

The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:04 +02:00
Kosuke Tatsukawa
694470273d ALSA: seq_oss: fix waitqueue_active without memory barrier in snd-seq-oss
snd_seq_oss_readq_put_event() seems to be missing a memory barrier which
might cause the waker to not notice the waiter and miss sending a
wake_up as in the following figure.

    snd_seq_oss_readq_put_event		    snd_seq_oss_readq_wait
------------------------------------------------------------------------
					/* wait_event_interruptible_timeout */
					 /* __wait_event_interruptible_timeout */
					  /* ___wait_event */
					  for (;;) {									 prepare_to_wait_event(&wq, &__wait,
					    state);
spin_lock_irqsave(&q->lock, flags);
if (waitqueue_active(&q->midi_sleep))
/* The CPU might reorder the test for
   the waitqueue up here, before
   prior writes complete */
					  if ((q->qlen>0 || q->head==q->tail)
					  ...
					  __ret = schedule_timeout(__ret)
if (q->qlen >= q->maxlen - 1) {
memcpy(&q->q[q->tail], ev, sizeof(*ev));
q->tail = (q->tail + 1) % q->maxlen;
q->qlen++;
------------------------------------------------------------------------

There are two other place in sound/core/seq/oss/ which have similar
code.  The attached patch removes the call to waitqueue_active() leaving
just wake_up() behind.  This fixes the problem because the call to
spin_lock_irqsave() in wake_up() will be an ACQUIRE operation.

I found this issue when I was looking through the linux source code
for places calling waitqueue_active() before wake_up*(), but without
preceding memory barriers, after sending a patch to fix a similar
issue in drivers/tty/n_tty.c  (Details about the original issue can be
found here: https://lkml.org/lkml/2015/9/28/849).

Signed-off-by: Kosuke Tatsukawa <tatsu@ab.jp.nec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:45:52 +02:00
Vinod Koul
70b4891cc8 ALSA: hda: make use of core codec fns
Now that we have introduced the core fns we should make hda use these
helpers

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-08 19:09:36 +02:00
Subhransu S. Prusty
1b5e6167c2 ALSA: hdac: Copy codec helpers to core
The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-08 19:09:30 +02:00
Takashi Iwai
601d62959d ASoC: Fixes for v4.3
Quite a few fixes here but they're all very small and driver specific,
 none of them really stand out if you aren't using the relevant hardware
 but they're all useful if you do happen to have an affected device.
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Merge tag 'asoc-fix-v4.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
2015-10-07 20:11:21 +02:00
Mark Brown
e4fc141d2a Merge remote-tracking branches 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8962' into asoc-linus 2015-10-07 16:07:50 +01:00
Mark Brown
1e2fa4cfdb Merge remote-tracking branches 'asoc/fix/db1200', 'asoc/fix/dwc', 'asoc/fix/imx-ssi', 'asoc/fix/maintainers', 'asoc/fix/rt5645', 'asoc/fix/sgtl5000' and 'asoc/fix/tas2552' into asoc-linus 2015-10-07 16:07:16 +01:00
Mark Brown
ff6afc8d0d MAINTAINERS: Remove wm97xx entry
Neither myself or Liam is especially interested in this driver any more
and the devices are already covered by the general ex-Wolfson entry so
just remove this.

Signed-off-by: Mark Brown <broonie@kernel.org>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
2015-10-06 12:21:17 +01:00
Andreas Dannenberg
e2600460bc ASoC: tas2552: fix dBscale-min declaration
The minimum volume level for the TAS2552 (control register value 0x00)
is -7dB however the driver declares it as -0.07dB.

Running amixer before the patch reports:
dBscale-min=-0.07dB,step=1.00dB,mute=0

Running amixer with the patch applied reports:
dBscale-min=-7.00dB,step=1.00dB,mute=0

Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-10-06 11:53:46 +01:00
Jeeja KP
a04267fd87 ALSA: hdac: Fix to check if stream not in use in release
if the stream is decoupled and both link and host are used, while
releasing the stream, need to check if link and host stream are
not in use. This patch adds fix to check if the host/link stream
is in used before coupling it back when releasing the stream.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 17:18:56 +02:00
Subhransu S. Prusty
88b19968a2 ALSA: hdac: Fix incorrect update of stream id mapping
Bits in LOSIDV need to be set to map the stream id for specific link.
Fixing this by setting the required bits in the register.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 17:18:44 +02:00
Takashi Iwai
225db5762d ALSA: synth: Fix conflicting OSS device registration on AWE32
When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel
warnings like:
  WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80()
  sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'

It's because both emux synth and opl3 drivers try to register their
OSS device object with the same static index number 0.  This hasn't
been a big problem until the recent rewrite of device management code
(that exposes sysfs at the same time), but it's been an obvious bug.

This patch works around it just by using a different index number of
emux synth object.  There can be a more elegant way to fix, but it's
enough for now, as this code won't be touched so often, in anyway.

Reported-and-tested-by: Michael Shell <list1@michaelshell.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 16:55:09 +02:00
Takashi Iwai
c7e1008048 ALSA: hda - Disable power_save_node for IDT 92HD73xx chips
The recent widget power saving introduced some unavoidable click
noises on old IDT 92HD73xx chips while it still seems working on the
compatible new chips.  In the bugzilla, we tried lots of tests and
workarounds, but they didn't help much.  So, let's disable the feature
for these specific chips as the least (but safest) fix.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-04 22:47:40 +02:00
Linus Torvalds
049e6dde7e Linux 4.3-rc4 2015-10-04 16:57:17 +01:00
Linus Torvalds
30c44659f4 Merge branch 'strscpy' of git://git.kernel.org/pub/scm/linux/kernel/git/cmetcalf/linux-tile
Pull strscpy string copy function implementation from Chris Metcalf.

Chris sent this during the merge window, but I waffled back and forth on
the pull request, which is why it's going in only now.

The new "strscpy()" function is definitely easier to use and more secure
than either strncpy() or strlcpy(), both of which are horrible nasty
interfaces that have serious and irredeemable problems.

strncpy() has a useless return value, and doesn't NUL-terminate an
overlong result.  To make matters worse, it pads a short result with
zeroes, which is a performance disaster if you have big buffers.

strlcpy(), by contrast, is a mis-designed "fix" for strlcpy(), lacking
the insane NUL padding, but having a differently broken return value
which returns the original length of the source string.  Which means
that it will read characters past the count from the source buffer, and
you have to trust the source to be properly terminated.  It also makes
error handling fragile, since the test for overflow is unnecessarily
subtle.

strscpy() avoids both these problems, guaranteeing the NUL termination
(but not excessive padding) if the destination size wasn't zero, and
making the overflow condition very obvious by returning -E2BIG.  It also
doesn't read past the size of the source, and can thus be used for
untrusted source data too.

So why did I waffle about this for so long?

Every time we introduce a new-and-improved interface, people start doing
these interminable series of trivial conversion patches.

And every time that happens, somebody does some silly mistake, and the
conversion patch to the improved interface actually makes things worse.
Because the patch is mindnumbing and trivial, nobody has the attention
span to look at it carefully, and it's usually done over large swatches
of source code which means that not every conversion gets tested.

So I'm pulling the strscpy() support because it *is* a better interface.
But I will refuse to pull mindless conversion patches.  Use this in
places where it makes sense, but don't do trivial patches to fix things
that aren't actually known to be broken.

* 'strscpy' of git://git.kernel.org/pub/scm/linux/kernel/git/cmetcalf/linux-tile:
  tile: use global strscpy() rather than private copy
  string: provide strscpy()
  Make asm/word-at-a-time.h available on all architectures
2015-10-04 16:31:13 +01:00
Takashi Sakamoto
425a570e1b ALSA: bebob: support Firewire I/O card of Mackie Onyx 1220/1620/1640
Current ALSA BeBoB drivers has an entry for this model, while the value of
vendor ID seems to be wrong according to an user's report.

The vendor had released no updated firmware, thus we can judge that this
model had not changed the content of its config ROM. It's reasonable to fix
the ID according to the report.

$ ./linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom

               ROM header and bus information block
               -----------------------------------------------------------------
400  0425720f  bus_info_length 4, crc_length 37, crc 29199
404  31333934  bus_name "1394"
408  f0646122  irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100,
               max_rec 6 (128), max_rom 1, gen 2, spd 2 (S400)
40c  00000ff2  company_id 00000f     |
410  00004697  device_id f200004697  | EUI-64 00000ff200004697

               root directory
               -----------------------------------------------------------------
414  000859be  directory_length 8, crc 22974
418  04000082  hardware version
41c  0c0083c0  node capabilities per IEEE 1394
420  03000ff2  vendor
424  8100000a  --> descriptor leaf at 44c
428  17010065  model
42c  8100000d  --> descriptor leaf at 460
430  13000910  version
434  d1000001  --> unit directory at 438

               unit directory at 438
               -----------------------------------------------------------------
438  0004ccec  directory_length 4, crc 52460
43c  1200a02d  specifier id: 1394 TA
440  13010001  version: AV/C
444  17010065  model
448  8100000d  --> descriptor leaf at 47c

               descriptor leaf at 44c
               -----------------------------------------------------------------
44c  0004152a  leaf_length 4, crc 5418
450  00000000  textual descriptor
454  00000000  minimal ASCII
458  4d61636b  "Mack"
45c  69650000  "ie"

               descriptor leaf at 460
               -----------------------------------------------------------------
460  000612b5  leaf_length 6, crc 4789
464  00000000  textual descriptor
468  00000000  minimal ASCII
46c  4f6e7978  "Onyx"
470  20466972  " Fir"
474  65776972  "ewir"
478  65000000  "e"

               descriptor leaf at 47c
               -----------------------------------------------------------------
47c  000612b5  leaf_length 6, crc 4789
480  00000000  textual descriptor
484  00000000  minimal ASCII
488  4f6e7978  "Onyx"
48c  20466972  " Fir"
490  65776972  "ewir"
494  65000000  "e"

$ cat /proc/asound/card3/firewire/firmware
Manufacturer:   bridgeCo
Protocol Ver:   1
Build Ver:      0
GUID:           0x00000FF200004697
Model ID:       0x82
Model Rev:      1
Firmware Date:  20040430
Firmware Time:  131527
Firmware ID:    0x10065
Firmware Ver:   2320
Base Addr:      0x20080000
Max Size:       1572864
Loader Date:    20040430
Loader Time:    112036

Reported-by: Andrzej Gansiniec <andrzej@gansiniec.pl>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-04 17:21:34 +02:00
Linus Torvalds
15ecf9a986 Assorted fixes for md in 4.3-rc
Two tagged for -stable
 One is really a cleanup to match and improve kmemcache interface.
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Merge tag 'md/4.3-fixes' of git://neil.brown.name/md

Pull md fixes from Neil Brown:
 "Assorted fixes for md in 4.3-rc.

  Two tagged for -stable, and one is really a cleanup to match and
  improve kmemcache interface.

* tag 'md/4.3-fixes' of git://neil.brown.name/md:
  md/bitmap: don't pass -1 to bitmap_storage_alloc.
  md/raid1: Avoid raid1 resync getting stuck
  md: drop null test before destroy functions
  md: clear CHANGE_PENDING in readonly array
  md/raid0: apply base queue limits *before* disk_stack_limits
  md/raid5: don't index beyond end of array in need_this_block().
  raid5: update analysis state for failed stripe
  md: wait for pending superblock updates before switching to read-only
2015-10-04 11:47:28 +01:00