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/20210607081250.13397-9-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/20210607081250.13397-8-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/20210607081250.13397-7-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/20210607081250.13397-6-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/20210607081250.13397-5-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/20210607081250.13397-4-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/20210607081250.13397-3-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 adds support for the hybrid model of MOTU Ultralite mk3 with
alpha connector, which is already discontinued. The hardware specification
of the model is the same as the one of FireWire-only model.
$ cd linux-firewire-utils
$ python3 src/crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 04101573 bus_info_length 4, crc_length 16, crc 5491
404 31333934 bus_name "1394"
408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
40c 0001f200 company_id 0001f2 |
410 000a059c device_id 00000a059c | EUI-64 0001f200000a059c
root directory
-----------------------------------------------------------------
414 0004ef04 directory_length 4, crc 61188
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 d1000002 --> unit directory at 428
424 8d000005 --> eui-64 leaf at 438
unit directory at 428
-----------------------------------------------------------------
428 0003f00b directory_length 3, crc 61451
42c 120001f2 specifier id
430 13000030 version
434 17103800 model
eui-64 leaf at 438
-----------------------------------------------------------------
438 0002d89c leaf_length 2, crc 55452
43c 0001f200 company_id 0001f2 |
440 000a059c device_id 00000a059c | EUI-64 0001f200000a059c
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210606043409.40019-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In all of drivers of ALSA firewire stack, the callback of .pointer and
.ack in snd_pcm_ops structure is done in acquired spin_lock of PCM
substream, therefore already under disabled kernel preemption.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210606025651.29970-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return -ENOMEM if kcalloc() fails. Currently the code returns success.
Fixes: f9e5ecdfc2 ("ALSA: firewire-lib: add replay target to cache sequence of packet")
Fixes: 6f24bb8a15 ("ALSA: firewire-lib: pool sequence of packet in IT context independently")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/YLtyL4VoArwVLor1@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the name string of several devices needing quirks to the Clevo-barebone
ones. Also make the names follow the same pattern for multiple Clevo names
referring to the same mainboard.
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Link: https://lore.kernel.org/r/20210604140207.8023-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code from the error handling
case instead of 0, as done elsewhere in this function.
Fixes: e50dfac81f ("ALSA: firewire-motu: cache event ticks in source packet header per data block")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Yang Yingliang <yangyingliang@huawei.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210603143203.582017-1-yangyingliang@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio driver behaves a bit strangely for the playback stream --
namely, it starts sending silent packets at PCM prepare state while
the actual data is submitted at first when the trigger START is kicked
off. This is a workaround for the behavior where URBs are processed
too quickly at the beginning. That is, if we start submitting URBs at
trigger START, the first few URBs will be immediately completed, and
this would result in the immediate period-elapsed calls right after
the start, which may confuse applications.
OTOH, submitting the data after silent URBs would, of course, result
in a certain delay of the actual data processing, and this is rather
more serious problem on modern systems, in practice.
This patch tries to revert the workaround and lets the URB submission
starting at PCM trigger for the playback again. As far as I've tested
with various backends (native ALSA, PA, JACK, PW), I haven't seen any
problems (famous last words :)
Note that the capture stream handling needs no such workaround, since
the capture is driven per received URB.
Link: https://lore.kernel.org/r/20210601162457.4877-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM delay accounting in USB-audio driver is a bit complex to
follow, and this is an attempt to improve the readability and provide
some potential fix.
Basically, the PCM position delay is calculated from two factors: the
in-flight data on URBs and the USB frame counter. For the playback
stream, we advance the hwptr already at submitting URBs. Those
"in-flight" data amount is now tracked, and this is used as the base
value for the PCM delay correction. The in-flight data is decreased
again at URB completion in return. For the capture stream, OTOH,
there is no in-flight data, hence the delay base is zero.
The USB frame counter is used in addition for correcting the current
position. The reference frame counter is updated at each submission
and receiving time, and the difference from the current counter value
is taken into account.
In this patch, each in-flight data bytes is recorded in the new
snd_usb_ctx.queued field, and the total in-flight amount is tracked in
snd_usb_substream.inflight_bytes field, as the replacement of
last_delay field.
Note that updating the hwptr after URB completion doesn't work for
PulseAudio who tries to scratch the buffer on the fly; USB-audio is
basically a double-buffer implementation, hence the scratching the
buffer can't work for the already submitted data. So we always update
hwptr beforehand. It's not ideal, but the delay account should give
enough correctness.
Link: https://lore.kernel.org/r/20210601162457.4877-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a bunch of lines calculating the buffer size in bytes at
each time. Keep the value in subs->buffer_bytes and use it
consistently for the code simplicity.
Link: https://lore.kernel.org/r/20210601162457.4877-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA firewire-motu driver to perform sequence replay for
media clock recovery.
Unlike the other types of device, the devices in MOTU FireWire series
require two levels of sequence replay; the sequence of the number of
data blocks per packet and the sequence of source packet header per data
block. The former is already cached by ALSA IEC 61883-1/6 packet streaming
engine and ready to be replayed. The latter is also cached by ALSA
firewire-motu driver itself with a previous patch. This commit takes
the driver to replay both of them from the caches.
The sequence replay is tested with below models:
* 828 mkII
* Traveler
* UltraLite
* 828 mk3 FireWire
* 828 mk3 Hybrid (except for high sampling transfer frequency
* UltraLite mk3 FireWire
* 4pre
* AudioExpress
Unfortunately, below models still don't generate better sound, requires
more work:
* 8pre
* 828 mk3 Hybrid at high sampling transfer frequency
As long as I know, MOTU protocol version 1 requires extra care of the
format of data block, thus below models are not supported yet in this
time:
* 828
* 896
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210602013406.26442-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices in MOTU FireWire series put source packet header (SPH) into
each data block of tx packet for presentation time of event. The format
of timestamp is compliant to IEC 61883-1, with cycle and offset fields
without sec field of 32 bit cycle time.
This commit takes ALSA firewire-motu driver to cache the presentation
time as offset from cycle in which the packet is transferred.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210602013406.26442-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA firewire-motu driver has some magic numbers from IEC 61883-1 to
operates source packet header (SPH). This commit replaces them with
macros.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210602013406.26442-2-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>
This commit takes ALSA dice driver to perform sequence replay for media
clock recovery.
Unlike the other types of device, DICE-based devices interpret the value
of syt field of CIP header in rx packets as presentation time for audio
playback, thus it's required for driver to compute value for outgoing
packet adequate to the device. It's done by media clock recovery by
handling tx packets.
The device starts packet transmission immediately at operation to
GLOBAL_ENABLE thus on-the-fly mode is not required.
DICE ASICs supports several pairs of isochronous packet streams.
Actually, maximum two pairs of streams are supported by devices.
We have three cases regarding to the number of streams:
1. a pair of streams
2. two tx packet streams and one rx packet streams
3. one tx packet streams and two rx packet streams
4. two pair of streams
The decision of playback timing is slightly different in the four cases.
In the case 1, sequence replay in the pair results in suitable playback
timing.
In the case 2, sequence replay from the first tx packet stream to rx
packet stream results in suitable playback timing.
In the case 3, sequence replay from tx packet stream to all of rx packet
stream results in suitable playback timing. Furthermore, the cycle to
start receiving packets should be the same between all rx packet streams.
In the case 4, sequence replay in each pair results in suitable playback
timing. Furthermore, the cycle to start receiving packets should be the
same between all rx packet streams.
The sequence replay is tested with below models:
* For case 1:
* TC Electronic Konnekt 24d (DiceII)
* TC Electronic Konnekt 8 (DiceII)
* TC Electronic Konnekt Live (DiceII)
* TC Electronic Impact Twin (DiceII)
* TC Electronic Digital Konnekt X32 (DiceII)
* TC Electronic Desktop Konnekt 6 (TCD2220)
* Solid State Logic Duende Classic (DiceII)
* Solid State Logic Duende Mini (DiceII)
* PreSonus FireStudio Project (TCD2210)
* PreSonus FireStudio Mobile (TCD2210)
* Lexicon I-ONIX FW810s (TCD2220)
* Avid Mbox 3 Pro (TCD2220)
* For case 2 (but case 1 depends on sampling transfer frequency):
* Alesis iO 26 (DiceII)
* Alesis iO 14 (DiceII)
* Alesis MultiMix 12 FireWire (DiceII)
* Focusrite Saffire Pro 26 (TCD2220)
* For case 3 (but case 1 depends on sampling transfer frequency):
* M-Audio Profire 610 (TCD2220)
* Loud Technology Mackie Onyx Blackbird (TCD2210)
* For case 4:
* TC Electronic Studio Konnekt 48 (DiceII + TCD2220)
* PreSonus FireStudio (DiceII)
* M-Audio Profire 2626 (TCD2220)
* Focusrite Liquid Saffire 56 (TCD2220)
* Focusrite Saffire Pro 40 (TCD2220)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210601081753.9191-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
NOTIFY_CLOCK_ACCEPTED notification is always generated as a result of
GLOBAL_CLOCK_SELECT operation, however NOTIFY_LOCK_CHG notification
doesn't, as long as the selected clock is already configured. In the case,
ALSA dice driver waits so long. It's inconvenient for some devices to lock
to the sequence of value in syt field of CIP header in rx packets.
This commit wait just for NOTIFY_CLOCK_ACCEPTED notification by reverting
changes partially done by two commits below:
* commit fbeac84dbe ("ALSA: dice: old firmware optimization for Dice notification")
* commit aec045b80d ("ALSA: dice: change notification mask to detect lock status change")
I note that the successful lock to the sequence of value in syt field of
CIP header in rx packets results in NOTIFY_EXT_STATUS notification, then
EXT_STATUS_ARX1_LOCKED bit stands in GLOBAL_EXTENDED_STATUS register.
The notification can occur enough after receiving the batch of rx packets.
When the sequence doesn't include value in syt field of CIP header in rx
packets adequate to the device, the notification occurs again and the bit
is off.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210601081753.9191-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA fireface driver to perform sequence replay for
media clock recovery.
The protocol specific to RME Fireface series is not compliant to
IEC 61883-1/6 since it has no CIP header, therefore presentation time
is not used for media clock recovery. The sequence of the number of data
blocks per packet is important.
I note that the device skips an isochronous cycle corresponding to an
empty packet or a NODATA packet in blocking transmission method of
IEC 61883-1/6. For sequence replay, the cycle is handled as receiving an
empty packet. Furthermore, it doesn't start packet transmission till
receiving any packet.
The sequence replay is tested with below models:
* Fireface 400
* Fireface 800
* Fireface 802
I note that it is better to initialize Fireface 400 in advance by
initialization transaction implemented in snd-fireface-ctl-service of
snd-firewire-ctl-services project. You can see whether initialized or
not by HOST LED on the device. Unless, the device often stops packet
transmission even if session starts.
I guess the sequence replay also works well with below models:
* Fireface UFX
* Fireface UCX
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA firewire-tascam driver to perform sequence replay
for media clock recovery.
The protocol specific to Tascam FireWire series is not compliant to
IEC 61883-1/6 in terms of syt field of CIP. The protocol doesn't use
presentation time in received CIP for playback timing. The sequence of
the number of data blocks per packet is important for media clock
recovery.
Although the devices in Tascam FireWire series transfer packets
regardless of receiving packets, the tx packets includes no events
in the beginning of streaming. It takes so long to multiplex any event
into the packet after receiving the sequence of packets. As long as I
experienced, it takes several thousands of isochronous cycle. Furthermore,
just after changing sampling transmission frequency, it stops multiplexing
event at once, then starts multiplexing again.
The sequence replay is tested with below models:
* FW-1884
* FW-1804
* FW-1082
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA firewire-digi00x driver to perform sequence replay
for media clock recovery.
All of models in Digidesign digi00x family don't transfer isochronous
packets till receiving isochronous packets. The on-the-fly mode is used
for the purpose. They don't interpret presentation time expressed in syt
field of received CIP, therefore the sequence of the number of data blocks
per packet is important for media clock recovery.
The sequence replay is tested with below models:
* Digidesign Digi 002
* Digidesign Digi 002 Rack
* Digidesign Digi 003
* Digidesign Digi 003 Rack
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit takes ALSA oxfw driver to perform sequence replay for media
clock recovery. Unfortunately, OXFW970 ASIC and its firmware has a quirk
called jumbo payload which skips several isochronous cycles for packet
transmission, thus the sequence replay is just adopted to OXFW971 ASIC.
As well as Fireworks, OXFW ASICs also ignores presentation time against
the way in IEC 61883-1/6.
The sequence replay is tested with below models:
* Tascam FireOne
* Stanton Magnetics SCS.1m
* Apogee Duet FireWire
For below models, the sequence replay is tested to be disabled:
* Griffin FireWave
* Behringer F-Control Audio 202
* Loud Technology Tapco Link.FireWire 4x6
* Loud Technology Mackie Onyx Satellite
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Echo Digital Audio Corporation had US patent US7599388B2 titled as
'System and method for high-bandwidth serial bus data transfer'. In the
patent, dual-banked shared memory is used to deliver data between
serial bus transmission and processor in FIFO way. The patent seems to be
used for Fireworks board module. The mechanism is not compliant to
synchronization based on presentation time expressed in syt field
of CIP header. Fireworks board module takes care of the sequence of
the number of data blocks per packet and just ignores the value of syt
field.
This commit takes fireworks driver to performs sequence replay for media
clock recovery. As long as I tested, Audiofire 2 and 4 have a quirk to
skip an isochronous cycle several thousands after starting packet
transmission.
The sequence replay is tested with below models:
* Loud Technology Mackie 400f
* Echo Audio Audiofire 12 (DSP model)
* Echo Audio Audiofire 12 (FPGA model)
* Echo Audio Audiofire 8 (DSP model)
* Echo Audio Audiofire 8 (FPGA model)
* Echo Audio Audiofire Pre8
* Echo Audio Audiofire 4
* Echo Audio Audiofire 2
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the design of Fireworks board module, the device does't adjust its
media clock voluntarily by the sequence of presentation time expressed in
syt field of CIP header of received packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210531025103.17880-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR,
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20210526121828.8460-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers of ALSA firewire stack can process packets for IT/IR context in
process context when the process operates ALSA PCM character device by
calling ioctl(2) with some requests. The ioctl requests are:
* SNDRV_PCM_IOCTL_HWSYNC
* SNDRV_PCM_IOCTL_SYNC_PTR
* SNDRV_PCM_IOCTL_REWIND
* SNDRV_PCM_IOCTL_FORWARD
* SNDRV_PCM_IOCTL_WRITEI_FRAMES
* SNDRV_PCM_IOCTL_READI_FRAMES
* SNDRV_PCM_IOCTL_WRITEN_FRAMES
* SNDRV_PCM_IOCTL_READN_FRAMES
This means that general application can process PCM frames apart from
hardware IRQ invocation, even if they are programmed by either IRQ-based
scheduling model or Timer-based scheduling model.
This commit add support for Timer-based scheduling model by allowing
PCM runtime to suppress both process wakeup per period and scheduling
hardware IRQ.
SNDRV_PCM_INFO_BATCH is obsoleted since ALSA IEC 61883-1/6 packet streaming
engine can report the number of transferred PCM frames within PCM period
boundary. The granularity equals to SYT_INTERVAL in blocking transmission.
In non-blocking transmission, it doesn't equal to SYT_INTERVAL but doesn't
exceed.
This patch is tested with PulseAudio, and --sched-model option of axfer
with fix against the issue reported at:
* https://lore.kernel.org/alsa-devel/687f9871-7484-1370-04d1-9c968e86f72b@linux.intel.com/#r
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527123253.174315-1-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>
ALSA IEC 61883-1/6 packet streaming engine uses pre-computed parameters
ideal for nominal sampling transfer frequency (STF) to transfer packets
to device since it was added 2011. As a result of user experience for a
decade, it is clear that the sequence is not suitable to some actual
devices. It takes the devices to generate noise, and causes any type of
discontinuity in the series of packet transferred from the device. It's
required for the engine to transfer packets according to effective STF.
The effective STF is given by media clock recovered by the sequence of
packet transferred from the target device. In the previous commit, the
sequence is already cached. The media clock recovery can be achieved by
analyzing the sequence.
In technological world, many ideas are proposed for media clock recovery.
However, the small part of them could be actually adopted in our case
since floating point arithmetic is not mostly available in Linux kernel
land.
This commit adopts the simple way from them; sequence replay, which means
that the sequence of parameters from incoming packet is used as is to
transfer outgoing packets. The media clock is not computed internally,
but the sequence of outgoing packet superficially looks to be generated by
the media clock.
The association between source and destination is decided when starting
AMDTP domain. When the target device supports a pair of isochronous packet
streams, the tx stream is source and the rx stream is destination. When it
supports two pair of streams, each of tx stream is associated to
corresponding rx stream in its order. When it supports less number of tx
streams than rx streams, the fist tx stream is selected for all of rx
streams. When it supports more tx streams than rx streams, the first tx
packet is associated to the rx stream.
As I noted in previous commit, the sequence of parameters from incoming
packet is different between devices, time to time. It is worse idea to
replay the sequence of parameters from a device for the sequence of
packet to the other devices even if they are in the same category of
device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210527122611.173711-3-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>
Don't populate the const array dsp_dma_stream_ids the stack but instead
make it static. Makes the object code smaller by 21 bytes.
Before:
text data bss dec hex filename
189012 70376 192 259580 3f5fc ./sound/pci/hda/patch_ca0132.o
After:
text data bss dec hex filename
188927 70440 192 259559 3f5e7 ./sound/pci/hda/patch_ca0132.o
(gcc version 10.3.0)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20210526160616.3764119-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse throws the following warning:
sound/pci/lx6464es/lx_core.c:677:34: error: self-comparison always
evaluates to false
This comparison and error message make no sense, let's remove them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse throws the following warning:
sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always
evaluates to false
This is likely a 16+ year old confusion between vp2 and vp.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20210526192957.449515-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR,
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210523071109.28940-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(),
which makes the code a bit shorter and easier to read.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20210524120007.39728-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pm_runtime_get_sync will increment pm usage counter even it failed.
Forgetting to putting operation will result in reference leak here.
Fix it by replacing it with pm_runtime_resume_and_get to keep usage
counter balanced.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Yufen Yu <yuyufen@huawei.com>
Link: https://lore.kernel.org/r/20210524093811.612302-1-yuyufen@huawei.com
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>
The member of callbacked in AMDTP stream structure is not used anymore.
Instead, ready_processing member is used to wake up yielding task of user
process.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices based on BeBoB ASICs or the devices in Tascam FireWire
series transfer a batch of NODATA packet or empty packet in the beginning
of packet streaming. To avoid processing them, current implementation uses
an option to skip processing content of tx packet during some initial
cycles. However, the hard-coded number is not enough useful.
This commit drops content of packets till the packet includes any event
firstly. The function of option is to skip processing content of tx packet
with any event after dropping.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210524031346.50539-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto
Dan Carpenter
Werner Sembach
Yang Yingliang
Takashi Iwai
Shaokun Zhang
YueHaibing
Colin Ian King
Pierre-Louis Bossart
zuoqilin
Yufen Yu