In the former commits, the callback of isochronous context runs on usual
work process. In the case, ALSA PCM device has a flag, nonatomic, to
acquire mutex lock instead of spin lock for PCM substream group.
This commit uses the flag. It has an advantage in the case that ALSA PCM
application uses the large size of intermediate buffer, since it takes
too long time even in tasklet softIRQ to process many of isochronous
packets, then result in the delay of system event due to disabled IRQ so
long. It is avertible to switch to nonatomic operation.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Edmund Raile <edmund.raile@protonmail.com>
Link: https://lore.kernel.org/r/20240904125155.461886-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Cc: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-8-tiwai@suse.de
The compile warnings at filling MIDI stream name strings are all
false-positive; the number of streams can't go so high.
For suppressing the warning, replace snprintf() with scnprintf().
As stated in the above, truncation doesn't matter.
Link: https://lore.kernel.org/r/20230915082802.28684-12-tiwai@suse.de
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change in ALSA core allows drivers to get the current PCM
state directly from runtime object. Replace the calls accordingly.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20220926135558.26580-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA bebob driver has duplicated entries for modalias of
'ieee1394:ven00000AACmo00000002sp0000A02Dver00010001' since entries for
two devices below have the same parameters:
* Acoustic Reality eAR Master One, Eroica, Figaro, and Ciaccona
* TerraTec Aureon 7.1 FireWire
I relied on FFADO revision 737 to add the former entry, on the other hand,
the latter is based on message posted by actual user with information of
sysfs node:
* https://sourceforge.net/p/ffado/mailman/ffado-user/thread/5743F969.2080204%40marcobaldo.ch/
It appears that they have OUI of Terratec Electronic GmbH (0x000aac) and
the same model ID, thus suffice to say that they have something common
in their internals.
Although it's not going to make a big difference, this commit arranges
the entries.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210705111455.63788-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user of FFADO project reported the issue of ToneWeal FW66. As a result,
the device is identified as one of applications of BeBoB solution.
I note that in the report the device returns contradictory result in plug
discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't
perform it thus it's likely to handle the device without issues.
I receive no reaction to test request for this patch yet, however it would
be worth to add support for it.
daniel@gibbonmoon:/sys/bus/firewire/devices/fw1$ grep -r . *
Binary file config_rom matches
dev:244:1
guid:0x0023270002000000
hardware_version:0x000002
is_local:0
model:0x020002
model_name:FW66
power/runtime_active_time:0
power/runtime_active_kids:0
power/runtime_usage:0
power/runtime_status:unsupported
power/async:disabled
power/runtime_suspended_time:0
power/runtime_enabled:disabled
power/control:auto
subsystem/drivers_autoprobe:1
uevent:MAJOR=244
uevent:MINOR=1
uevent:DEVNAME=fw1
units:0x00a02d:0x010001
vendor:0x002327
vendor_name:ToneWeal
fw1.0/uevent:MODALIAS=ieee1394:ven00002327mo00020002sp0000A02Dver00010001
fw1.0/power/runtime_active_time:0
fw1.0/power/runtime_active_kids:0
fw1.0/power/runtime_usage:0
fw1.0/power/runtime_status:unsupported
fw1.0/power/async:disabled
fw1.0/power/runtime_suspended_time:0
fw1.0/power/runtime_enabled:disabled
fw1.0/power/control:auto
fw1.0/model:0x020002
fw1.0/rom_index:15
fw1.0/specifier_id:0x00a02d
fw1.0/model_name:FW66
fw1.0/version:0x010001
fw1.0/modalias:ieee1394:ven00002327mo00020002sp0000A02Dver00010001
Cc: Daniel Jozsef <daniel.jozsef@gmail.com>
Reference: https://lore.kernel.org/alsa-devel/20200119164335.GA11974@workstation/
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210619083922.16060-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Below devices reports zero as the number of channels for external output
plug with MIDI type:
* Yamaha GO44/GO46
* Terratec Phase 24/X24
As a result, rx packet format is invalid and they generate silent sound.
This is a regression added in v5.13.
This commit fixes the bug, addressed at a commit 1bd1b3be86 ("ALSA:
bebob: perform sequence replay for media clock recovery").
Cc: <stable@vger.kernel.org>
Fixes: 5c6ea94f2b ("ALSA: bebob: detect the number of available MIDI ports")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210618040447.113306-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirking bit-flags are currently set as contiguous integer enum values
and so currently SND_BEBOB_QUIRK_INITIAL_DISCONTINUOUS_DBC is zero and so
he quirking never getting set or tested correctly for this quirk. Fix this
by setting the quirking constants as shifted bit values.
Addresses-Coverity: ("Bitwise-and with zero")
Fixes: 93cd12d6e8 ("ALSA: bebob: code refactoring for model-dependent quirks")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210615142048.59900-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For M-Audio FireWire 410, the value of immediate entry for vendor in unit
directory is the value for BridgeCo. AG OUI. It seems that M-Audio uses
initial settings as is for its product.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210611093730.78254-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Mackie D.2 FireWire option card, 0x00000f is used for the value of
immediate entry for vendor in unit directory. The value comes from report
by FFADO user in below page:
* http://subversion.ffado.org/wiki/AvcModels/MackieD.2.
However, it seems to be wrong. There are two causes; vendor's mistake to
decide value for GUID field in configuration ROM against standard, as
Stefan Richter mentioned in below post:
* https://lore.kernel.org/alsa-devel/1443917823-13516-1-git-send-email-o-takashi@sakamocchi.jp/#t
Another is implementation of libffado. The library doesn't print out the
value from immediate entry for vendor in unit directory. It just print out
the first 6 bytes of GUID as vendor ID.
This commit replaces with correct vendor OUI.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210611093730.78254-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although unit directory in root directory of configuration ROM has the
same value (0x00a02d) for its specifier_id entry to express AV/C device,
it has two cases for the value (0x010001/0x014001) to version entry.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210611093730.78254-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a commit c4d860a0d2 ("ALSA: bebob: loosen up severity of checking
continuity for BeBoB v3 quirk"), a workaround was added for the quirk in
BeBoB protocol version 3 against the discontinuity of data block counter.
As long as seeing with sequence replay for media clock recovery, such
quirk disappears.
This commit deletes the workaround.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210611035003.26852-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a commit d3eabe939a ("ALSA: bebob: expand sleep just after breaking
connections for protocol version 1"), a workaround was added for a quirk
of freeze in BeBoB protocol version 1. As long as seeing with sequence
replay for media clock recovery, the quirk disappears.
This commit removes the workaround.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210611035003.26852-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The delayed registration of sound card instance brings less benefit than
complication of kobject management. This commit ceases from it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210607081250.13397-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA bebob driver to perform sequence replay for media
clock recovery.
Many users have reported discontinuity of data block counter field of CIP
header in tx packet from the devices based on BeBoB ASICs. In the worst
case, the device corrupts not to respond to any transaction, then generate
bus-reset voluntarily for recovery. The sequence replay for media clock
recovery is expected to suppress most of the problems.
In the beginning of packet streaming, the device transfers NODATA packets
for a while, then multiplexes any event and syt information. ALSA
IEC 61883-1/6 packet streaming engine has implementation for it to drop
the initial NODATA packets. It starts sequence replay when detecting any
event multiplexed to tx packets.
The sequence replay is tested with below models:
* Focusrite Saffire
* Focusrite Saffire LE
* Focusrite Saffire Pro 10 I/O
* Focusrite Saffire Pro 26 I/O
* M-Audio FireWire Solo
* M-Audio FireWire Audiophile
* M-Audio Ozonic
* M-Audio FireWire 410
* M-Audio FireWire 1814
* Edirol FA-66
* ESI Quatafire 610
* Apogee Ensemble
* Phonic Firefly 202
* Behringer F-Control Audio 610
Unfortunately, below models doesn't generate sound. This seems regression
introduced recent few years:
* Stanton Final Scratch ScratchAmp at middle sampling transfer frequency
* Yamaha GO44
* Yamaha GO46
* Terratec Phase x24
As I reported, below model has quirk of discontinuity:
* M-Audio ProFire Lightbridge
DM1000/DM1100 ASICs in BeBoB solution are known to have bugs at switch of
sampling transfer frequency between low/middle/high rates. The switch
generates the similar problems about which I mention in the above. Some
vendors customizes firmware so that the switch of frequency is done in
vendor-specific registers, then restrict users to switch the frequency.
For example of Focusrite Saffire Pro 10 i/o and 26 i/o, users allows to
switch the frequency within the three steps; e.g. 44.1/48.0 kHz are
available at low step. Between the steps, extra operation is required and
it always generates bus-reset.
Another example of Edirol FA-66, users are prohibited to switch the
frequency by software. It's done by hardware switch and power-off.
I note that the sequence replay is not a solution for the ASIC bugs. Users
need to disconnect the device corrupted by the bug, then reconnect it to
refresh state machine inner the ASIC.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210601081753.9191-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Models in below series start transmission of packet after receiving the
sequence of packets:
* Digidesign Digi00x family
* RME Fireface series
Additionally, models in Tascam FireWire series start multiplexing PCM
frames into packets enough after receiving packets. It's required to
transfer packets on-the-fly for the above models according to nominal
sampling transfer frequency before starting sequence replay.
This commit allows drivers to decide whether the engine transfers packet
on-the-fly or not.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In design of audio and music unit in IEEE 1394 bus, feedback of
effective sampling transfer frequency (STF) is delivered by packets
transferred from device. The devices supported by ALSA firewire stack
are categorized to three groups regarding to it.
* Group 1:
* Echo Audio Fireworks board module
* Oxford Semiconductor OXFW971 ASIC
* Digidesign Digi00x family
* Tascam FireWire series
* RME Fireface series
* Group 2:
* BridgeCo. DM1000/DM1100/DM1500 ASICs for BeBoB solution
* TC Applied Technologies DICE ASICs
* Group 3:
* Mark of the Unicord FireWire series
In group 1, the effective STF is determined by the sequence of the number
of events per packet. In group 2, the sequence of presentation timestamp
expressed in syt field of CIP header is interpreted as well. In group 3,
the presentation timestamp is expressed in source packet header (SPH) of
each data block.
I note that some models doesn't take care of effective STF with large
internal buffer. It's reasonable to name it as group 0:
* Group 0
* Oxford Semiconductor OXFW970 ASIC
The effective STF is known to be slightly different from nominal STF for
all of devices, and to be different between the devices. Furthermore, the
effective STF is known to be shifted for long-period transmission. This
makes it hard for software to satisfy the effective STF when processing
packets to the device.
The effective STF is deterministic as a result of analyzing the batch of
packet transferred from the device. For the analysis, caching the sequence
of parameter in the packet is required.
This commit adds an option so that AMDTP domain structure takes AMDTP
stream structure to cache the sequence of parameters in packet transferred
from the device. The parameters are offset ticks of syt field against the
cycle to receive the packet and the number of data blocks per packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, ALSA IEC 61883-1/6 packet streaming engine drops
initial tx packets till the packet includes any event. This allows ALSA
bebob driver not to give option to skip initial packet since the engine
does drop the initial packet.
However, M-Audio ProFire Lightbridge has a quirk to stop packet
transmission after start multiplexing event to the packet. After several
thousands cycles, it restart packet transmission again.
This commit specializes the usage of initial skip option for the model.
Additionally, this commit expands timeout enough to wait processing
content of tx packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When starting AMDTP domain, tasks in process context yields running CPU
till all of isochronous context get callback, with an assumption that
it's OK to process content of packet.
However several isochronous cycles are skipped to transfer rx packets, or
the content of rx packets are dropped, to manage the timing to start
processing the packets.
This commit changes the timing for tasks in process context to wake up
when processing content of packet is actually ready.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210520040154.80450-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current implementation of ALSA IEC 61883-1/6 packet streaming engine
allows drivers to decide isochronous cycle to start IR context. This
option is mainly used to avoid processing the sequence of packet with
some quirks; e.g. discontinuity of counter. However, it's inconvenient
to fail to continue packet processing when the target device doesn't
start transmission of packet till the decided cycle.
This commit changes the behaviour. As an alternative to the start cycle
for IR context, the cycle count to drop content of packet in the beginning
of IR context.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210520040154.80450-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 0edabdfe89.
I've explained that optional FireWire card for d.2 is also built-in to
d.2 Pro, however it's wrong. The optional card uses DM1000 ASIC and has
'Mackie DJ Mixer' in its model name of configuration ROM. On the other
hand, built-in FireWire card for d.2 Pro and d.4 Pro uses OXFW971 ASIC
and has 'd.Pro' in its model name according to manuals and user
experiences. The former card is not the card for d.2 Pro. They are similar
in appearance but different internally.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518084557.102681-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mackie d.2 has an extension card for IEEE 1394 communication, which uses
BridgeCo DM1000 ASIC. On the other hand, Mackie d.4 Pro has built-in
function for IEEE 1394 communication by Oxford Semiconductor OXFW971,
according to schematic diagram available in Mackie website. Although I
misunderstood that Mackie d.2 Pro would be also a model with OXFW971,
it's wrong. Mackie d.2 Pro is a model which includes the extension card
as factory settings.
This commit fixes entries in Kconfig and comment in ALSA OXFW driver.
Cc: <stable@vger.kernel.org>
Fixes: fd6f4b0dc1 ("ALSA: bebob: Add skelton for BeBoB based devices")
Fixes: ec4dba5053 ("ALSA: oxfw: Add support for Behringer/Mackie devices")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current implementation of bebob driver doesn't correctly handle the case
that the device has multiple MIDI ports. The cause is the number of MIDI
conformant data channels is passed to AM824 data block processing layer.
This commit fixes the bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210321032831.340278-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current implementation counts the number of input/output plugs for MIDI
type and uses the count as the number of physical MIDI ports. However,
the number of channels of the port represents the count.
This commit fixes the bug by additional vendor-specific AVC command
extension.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210321032831.340278-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA bebob driver scans supported formats of packet for each direction
when probing the target device. Some helper functions are used for the
scanning, however its implementation is not necessarily irredundant.
This commit refactors the helper functions to remove redundant codes.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210321032831.340278-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers in ALSA firewire stack supports eventing to userspace
applications via ALSA hwdep interface. All of the drivers supports stream
lock events. Some of them supports their unique events according to
specification of target device.
ALSA bebob driver supports the stream lock event only. In the case, it's
enough to check condition only in loop with process blocking. However,
current implementation check it again after breaking the loop.
This commit removes the redundant check.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Reported-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210125140208.26318-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
strlcpy is deprecated. see: Documentation/process/deprecated.rst
Change the calls that do not use the strlcpy return value to the
preferred strscpy.
Done with cocci script:
@@
expression e1, e2, e3;
@@
- strlcpy(
+ strscpy(
e1, e2, e3);
This cocci script leaves the instances where the return value is
used unchanged.
After this patch, sound/ has 3 uses of strlcpy() that need to be
manually inspected for conversion and changed one day.
$ git grep -w strlcpy sound/
sound/usb/card.c: len = strlcpy(card->longname, s, sizeof(card->longname));
sound/usb/mixer.c: return strlcpy(buf, p->name, buflen);
sound/usb/mixer.c: return strlcpy(buf, p->names[index], buflen);
Miscellenea:
o Remove trailing whitespace in conversion of sound/core/hwdep.c
Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "count" variable needs to be capped on every path so that we don't
copy too much information to the user.
Fixes: 618eabeae7 ("ALSA: bebob: Add hwdep interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201007074928.GA2529578@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
KBUILD_MODNAME is available to name kernel modules according to its object
name. This commit uses the macro instead of string for name field of
struct driver since drivers in ALSA firewire stack have the same name of
each object name.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200306135229.11659-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All the PCM ioctl ops of ALSA FireWire drivers do nothing but calling
the default handler.
Now PCM core accepts NULL as the default ioctl ops(*), so let's drop
altogether.
(*) commit fc033cbf6f ("ALSA: pcm: Allow NULL ioctl ops")
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191210061145.24641-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the drivers with the new managed buffer allocation API.
The superfluous snd_pcm_lib_malloc_pages() and
snd_pcm_lib_free_pages() calls are dropped.
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191209192422.23902-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change (*) in the ALSA memalloc core allows us to drop the
special vmalloc-specific allocation and page handling. This patch
coverts to the common code.
(*) 1fe7f397cf: ALSA: memalloc: Add vmalloc buffer allocation
support
7e8edae39f: ALSA: pcm: Handle special page mapping in the
default mmap handler
Link: https://lore.kernel.org/r/20191105151856.10785-11-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Focusrite Saffire Pro i/o, the lowest 8 bits of register represents
configured source of sampling clock. The next lowest 8 bits represents
whether the configured source is actually detected or not just after
the register is changed for the source.
Current implementation evaluates whole the register to detect configured
source. This results in failure due to the next lowest 8 bits when the
source is connected in advance.
This commit fixes the bug.
Fixes: 25784ec2d0 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series")
Cc: <stable@vger.kernel.org> # v3.16+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191102150920.20367-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally BeBeB ASICs and firmware supports clock mode to synchronizing
to syt field of received isoc packet. This mode is known as 'SYT Match'
slightly described in IEC 61883-6 (but no detail mechanisms). In this
mode, drivers can control sampling clock in device. Driver for Windows
and macOS uses this feature to perform synchronization for devices
on the same bus.
In this mode, a plug of Music subunit for synchronization is connected
to a plug of isoc unit for incoming packet streaming, then the order to
establish connections is INPUT_PLUG first, OUTPUT_PLUG second.
This commit implements the above.
Actually each device works with its own clock for sampling, therefore
the original design is hardly implemented to vendor's products.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I investigated, there's some cases about the delay for device
between establishing OUTPUT_PLUG and transmitting first isoc packet. For
devices which support BeBoB protocol version 1 can transmit the packet
within several hundred milliseconds, while for devices which support
BeBoB protocol version 3 can transmit the packet within 2 seconds.
Devices with protocol version 1:
* Edirol FA-66
* Yamaha GO46
* Terratec Phase x24 FW
* M-Audio FireWire AudioPhile
* M-Audio FireWire Solo
* M-Audio FireWire 1814
* M-Audio FireWire 410
* Focusrite Saffire Pro 26 I/O
Devices with protocol version 3:
* M-Audio Profire Lightbridge
* Behringer FCA610
* Phonic Firefly 202
At present ALSA bebob driver postpones starting IR context during
1.5 sec for all of supported devices. The delay is too long for
devices with protocol version 1, while it's not enough for devices with
protocol version 3.
This commit improves the delay for these protocols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as I investigated, some devices with BeBoB protocol version 1
can be freezed during several hundreds milliseconds after breaking
connections. When accessing during the freezed time, any transaction
is corrupted. In the worst case, the device is going to reboot.
I can see this issue in:
* Roland FA-66
* M-Audio FireWire Solo
This commit expands sleep just after breaking connections to avoid
the freezed time as much as possible. I note that the freeze/reboot
behaviour is similar to below models:
* Focusrite Saffire Pro 10 I/O
* Focusrite Saffire Pro 26 I/O
The above models certainly reboot after breaking connections.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191101131323.17300-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A helper function of ALSA bebob driver returns negative value in a
function which has a prototype to return unsigned value.
This commit fixes it by changing the prototype.
Fixes: eb7b3a056c ("ALSA: bebob: Add commands and connections/streams management")
Cc: <stable@vger.kernel.org> # v3.16+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191026030620.12077-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have a quirk to postpone transmission of isoc packet for
several dozen or hundred isoc cycles since configured to transmit.
Furthermore, some devices have a quirk to transmit isoc packet with
discontinued data of its header.
In 1394 OHCI specification, software allows to start isoc context with
certain isoc cycle. Linux firewire subsystem has kernel API to use it
as well.
This commit uses the functionality of 1394 OHCI controller to handle
the quirks. At present, this feature is convenient to ALSA bebob and
fireface driver. As a result, some devices can be safely handled, as
long as I know:
- MAudio FireWire solo
- MAudio ProFire Lightbridge
- MAudio FireWire 410
- Roland FA-66
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An isoc context for AMDTP stream is flushed to queue packet
by a call of pcm.ack. This commit extends this for AMDTP
domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An isoc context for AMDTP stream is flushed to queue packet
by a call of pcm.pointer. This commit extends this for AMDTP
domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA bebob driver to share PCM buffer size for both
capture and playback PCM substream. When AMDTP domain starts for one
of the PCM substream, buffer size of the PCM substream is stores to
AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of packets in packet buffer has been fixed number (=48) since
first commit of ALSA IEC 61883-1/6 packet streaming engine.
This commit allows the engine to use variable number of packets in the
buffer. The size is calculated by a parameter in AMDTP domain structure
surely to store the number of events in the packets of buffer. Although
the value of parameter is expected to come from 'period size' parameter
of PCM substream, at present 48 is still used.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-11-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
