*-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>
Hard-coded stream format parameters are added for Weiss Engineering
FireWire devices. When the device vendor and model match, the parameters
are copied into the stream format cache. This allows for setting all
supported sampling rates up to 192kHz, and consequently adjusting the
number of available I/O channels.
Signed-off-by: Rolf Anderegg <rolf.anderegg@weiss.ch>
Signed-off-by: Michele Perrone <michele.perrone@weiss.ch>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20230809002631.750120-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This looks like a relatively calm development cycle; there have been
only few changes in ALSA and ASoC core sides while we get lots of
device-specific fixes and updates as usual. Most of commits are about
ASoC, including Intel SOF/AVS and many device tree updates.
Below are some highlights:
Core:
- Improvement in memalloc helper for fallback allocations
- More cleanups of ASoC DAPM code
ASoC:
- Factoring out of mapping hw_params onto SoundWire configuration
- The ever ongoing overhauls of the Intel DSP code continue, including
support for loading libraries and probes with IPC4 on SOF.
- Support for more sample formats on JZ4740
- Lots of device tree conversions and fixups
- Support for Allwinner D1, a range of AMD and Intel systems, Mediatek
systems with multiple DMICs, Nuvoton NAU8318, NXP fsl_rpmsg and
i.MX93, Qualcomm AudioReach Enable, MFC and SAL, RealTek RT1318 and
Rockchip RK3588
ALSA:
- Addition of PCM kselftest; still minimalistic but can be extended
in future
- Fixes for corner-case XRUNs with USB-audio implicit feedback mode
- Usual device-specific quirk updates for USB- and HD-audio
- FireWire DICE updates
Also, this PR also contains a few cross-tree updates:
- Some OMAP board file updates for removal of relevant OMAP platforms
- A new I2C API update for I2C probe API adaption
- A DRM update for the further hdmi-codec updates
-----BEGIN PGP SIGNATURE-----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=QV3a
-----END PGP SIGNATURE-----
Merge tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This looks like a relatively calm development cycle; there have been
only few changes in ALSA and ASoC core sides while we get lots of
device-specific fixes and updates as usual. Most of commits are about
ASoC, including Intel SOF/AVS and many device tree updates.
Below are some highlights:
Core:
- Improvement in memalloc helper for fallback allocations
- More cleanups of ASoC DAPM code
ASoC:
- Factoring out of mapping hw_params onto SoundWire configuration
- The ever ongoing overhauls of the Intel DSP code continue,
including support for loading libraries and probes with IPC4 on
SOF.
- Support for more sample formats on JZ4740
- Lots of device tree conversions and fixups
- Support for Allwinner D1, a range of AMD and Intel systems,
Mediatek systems with multiple DMICs, Nuvoton NAU8318, NXP
fsl_rpmsg and i.MX93, Qualcomm AudioReach Enable, MFC and SAL,
RealTek RT1318 and Rockchip RK3588
ALSA:
- Addition of PCM kselftest; still minimalistic but can be extended
in future
- Fixes for corner-case XRUNs with USB-audio implicit feedback mode
- Usual device-specific quirk updates for USB- and HD-audio
- FireWire DICE updates
This also contains a few cross-tree updates:
- Some OMAP board file updates for removal of relevant OMAP platforms
- A new I2C API update for I2C probe API adaption
- A DRM update for the further hdmi-codec updates"
* tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (417 commits)
ALSA: mts64: fix possible null-ptr-defer in snd_mts64_interrupt
ALSA: patch_realtek: Fix Dell Inspiron Plus 16
ALSA: hda/cirrus: Add extra 10 ms delay to allow PLL settle and lock.
ASoC: dt-bindings: Correct Alexandre Belloni email
ASoC: dt-bindings: maxim,max98504: Convert to DT schema
ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
ASoC: dt-bindings: Reference common DAI properties
ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
ASoC: uniphier: aio-core: Make some read-only arrays static const
ASoC: wcd938x: Make read-only array minCode_param static const
ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
ASoC : SOF: amd: Add support for IPC and DSP dumps
ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
ALSA: usb-audio: Workaround for XRUN at prepare
ALSA: pcm: Handle XRUN at trigger START
ALSA: pcm: Set missing stop_operating flag at undoing trigger start
drm: tda99x: Don't advertise non-existent capture support
ASoC: hdmi-codec: Allow playback and capture to be disabled
kselftest/alsa: Add more coverage of sample rates and channel counts
...
Following a commit 1dd0dd0b1f ("ALSA: firewire: Remove some left-over
license text in sound/firewire"), this patch removes it added carelessly.
Fixes: 2133dc91d6 ("ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221201030100.31495-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TC Applied Technologies (TCAT) produces TCD3070 as final DICE ASIC for
communication in IEEE 1394 bus for IEC 61883-1/6 protocol. As long as I
know, latter model of Focusrite Saffire Pro 40 is an application of the
ASIC and only in the market for consumers.
This patchset adds support for the device. The device has several
remarkable points.
1. No support for extended synchronization information section in TCAT
general protocol. The value of GLOBAL_EXTENDED_STATUS register is always
zero. Additionally, NOTIFY_EXT_STATUS message is never emitted.
2. No support for TCAT protocol extension. Hard coding is required for
format of CIP payload.
3. During several seconds after changing sampling rate, the block to
process PCM frames is under disfunction. When starting packet streaming
during the state, the block is never function till configuring different
sampling rate and several seconds.
This commit adds support for the device. The item 1 and 2 can be
adaptable, while item 3 is not. It's not preferable that user process
is forced to sleep during the disfunction in the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS or SNDRV_PCM_IOCTL_PREPARE request. It's
inconvenient but let user configure preferable sampling rate in advance
of starting PCM substream.
The content of configuration ROM in the device I used is available at:
* https://github.com/takaswie/am-config-roms/
I note that any mixer control operation is implemented by unique
transaction. The frame of request consists of 16 bytes header followed
by payload.
header (4 quadlets):
1st: the type of request, prefixed with 0x8000
2nd: counter at 2 bytes in MSB side, the length of data at 2 bytes in LSB
side
3rd: parameter 0
4th: parameter 1
payload (variable length if need):
5th-: data according to parameters
The request frame is sent by block write request to 0x'ffff'e040'01c0.
The frame of response is similar to the frame of request, but it is
header only, thus fixed to 16 bytes. The response frame is sent to the
address which is registered by lock transaction to 0x'ffff'e040'0008.
If the operation results in batch of data, the 2nd quadlet of header
includes the length of data like request. The data is itself readable
by read block request to 0x'ffff'e040'0030, which includes both
header and payload for data, thus the length to read should be the
length of data plus 16 bytes for header
The actual value of request, parameter 0, parameter 1, and data is
unclear yet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221130143313.43880-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Lexicon I-ONIX FW810S, the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS can returns -ETIMEDOUT. This is a regression due
to the commit 41319eb56e ("ALSA: dice: wait just for
NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation"). The device
does not emit NOTIFY_CLOCK_ACCEPTED notification when accepting
GLOBAL_CLOCK_SELECT operation with the same parameters as current ones.
This commit fixes the regression. When receiving no notification, return
-ETIMEDOUT as long as operating for any change.
Fixes: 41319eb56e ("ALSA: dice: wait just for NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221130130604.29774-1-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>
There is already a SPDX-License-Identifier tag, so the corresponding license
text can be removed.
While at it, be more consistent and:
- add a missing .c (ff-protocol-latter)
- remove an empty line (motu-protocol-v1)
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/2bfe76c7eeb0f5205a1427e280bf8d9da0354a62.1664110649.git.christophe.jaillet@wanadoo.fr
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>
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>
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>
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>
At high sampling transfer frequency, TC Electronic Konnekt Live
transfers/receives 6 audio data frames in multi bit linear audio data
channel of data block in CIP payload. Current hard-coded stream format
is wrong.
Cc: <stable@vger.kernel.org>
Fixes: f1f0f330b1 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518012612.37268-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA dice driver detects jumbo payload at high sampling transfer frequency
for below models:
* Avid M-Box 3 Pro
* M-Audio Profire 610
* M-Audio Profire 2626
Although many DICE-based devices have a quirk at high sampling transfer
frequency to multiplex double number of PCM frames into data block than
the number in IEC 61883-1/6, the above devices are just compliant to
IEC 61883-1/6.
This commit disables the mode of double_pcm_frames for the models.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210518012510.37126-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alesis iO 26 FireWire has two pairs of digital optical interface. It
delivers PCM frames from the interfaces by second isochronous packet
streaming. Although both of the interfaces are available at 44.1/48.0
kHz, first one of them is only available at 88.2/96.0 kHz. It reduces
the number of PCM samples to 4 in Multi Bit Linear Audio data channel
of data blocks on the second isochronous packet streaming.
This commit fixes hardcoded stream formats.
Cc: <stable@vger.kernel.org>
Fixes: 28b208f600 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210513125652.110249-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When node is removed from IEEE 1394 bus, any transaction fails to the node.
In the case, ALSA dice driver doesn't stop isochronous contexts even if
they are running. As a result, null pointer dereference occurs in callback
from the running context.
This commit fixes the bug to release isochronous contexts always.
Cc: <stable@vger.kernel.org> # v5.4 or later
Fixes: e9f21129b8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210312093407.23437-1-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>
ALSA dice driver expects devices to multiplex MIDI messages into first
port of isochronous communication. Actually devices perform for it.
However, check of stream format is invalid for second port of isochronous
communication. As a result, when the device supports two ports for
isochronous communication and the stream format is hard-coded, ALSA
dice driver fails to start packet streaming.
This commit loosens stream format check for MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113084630.14305-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At failure of attempt to detect protocol extension, ALSA dice driver
should be fallback to limited functionality. However it's not.
This commit fixes it.
Cc: <stable@vger.kernel.org> # v4.18+
Fixes: 58579c056c ("ALSA: dice: use extended protocol to detect available stream formats")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113084630.14305-2-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>
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 dice 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/20191017155424.885-6-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.
Dice hardware has a quirk called as 'Dual Wire'. For a case of higher
sampling transmission frequency, this commit performs calculations between
the number of PCM frames and the number of events in AMDTP stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-14-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data
channels per data block in CIP.
Both iO14 and iO26 have the same contents in their configuration ROM.
For this reason, ALSA Dice driver attempts to distinguish them according
to the value of TX0_AUDIO register at probe callback. Although the way is
valid at lower and middle sampling rate, it's lastly invalid at higher
sampling rate because because the two models returns the same value for
read transaction to the register.
In the most cases, users just plug-in the device and ALSA dice driver
detects it. In the case, the device runs at lower sampling rate and
the driver detects expectedly. For this reason, this commit leaves the
way to detect as is.
Fixes: 28b208f600 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Cc: <stable@vger.kernel.org> # v4.18+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190916101851.30409-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When introducing AMDTP domain to ALSA dice driver, error path does not
handle error correctly. This commit fixes the bug.
Fixes: e9f21129b8 ("ALSA: dice: support AMDTP domain")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.
- Lots more work on moving towards a component based framework from
Morimoto-san.
- Support for force disconnecting muxes from Jerome Brunet.
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl0jGwUTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0LD4B/9AkutfS+vznOrk0V0wFb2SUfjwE4Pr
+z/kAehohAOl/7pg9Dun/lmZYBWMyOM2aYmK81ahEo2DfO+uzwkwjCaXFjGVGwEK
j7XpWkrIjKnou/z1FeALgVvt+crzdy5iNWC04AbKaP2WHCcI7zvPQIsBta/V0OJt
lg+j0J7pagnTMcgV1+qJdaASmofy/hpoZ79Gv0PIfGC8hpJ/3mBgcNPCLQrJtD4R
v+tzvCZNrZVqCanwLf3vouEm1bpWYOpI+Wdmu4u6rY7MhmCj72EJ2zyfdm/qtaxF
e7whgCyOQFkWe7NgDn0G08aAT6LsaxOtPNr7H8tL8S8sw8425fqeOouV
=n/HQ
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.3
This is a very big update, mainly thanks to Morimoto-san's refactoring
work and some fairly large new drivers.
- Lots more work on moving towards a component based framework from
Morimoto-san.
- Support for force disconnecting muxes from Jerome Brunet.
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, several types of sampling data can be multiplexed into
payload of common isochronous packet (CIP). For typical audio and music
units, PCM samples and MIDI messages are multiplexed into one packet
streaming.
ALSA dice driver allows applications of rawmidi interface to start
packet streaming for transmission of MIDI messages. However at error
path, the reference count of stream functionality is not operated
correctly. This can brings a bug that packet streaming is not stopped
when all referrers release the count.
This commit fixes the bug.
Fixes: 3cd2c2d780 ("ALSA: dice: reserve/release isochronous resources in pcm.hw_params/hw_free callbacks")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
>From callbacks for pcm and rawmidi interfaces, the functions to stop
and release duplex streams are called at the same time. This commit
merges the two functions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit fixes the warning due to returning uninitialized value
from start_streams() helper function.
sound/firewire/dice/dice-stream.c: In function 'start_streams.isra.0':
>> sound/firewire/dice/dice-stream.c:350:6: warning: 'err' may be used uninitialized in this function [-Wmaybe-uninitialized]
int err;
^~~
Reported-by: kbuild test robot <lkp@intel.com>
Fixes: 3cd2c2d780 ("ALSA: dice: reserve/release isochronous resources in pcm.hw_params/hw_free callbacks")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pairs of pcm.hw_params callbacks and .hw_free callbacks for both
direction have no differences.
This commit unifies the pairs.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After bus reset, isochronous resource manager releases all of allocated
isochronous resources. The nodes to transfer isochronous packet should
request reallocation of the resources.
However, between the bus-reset and invocation of 'struct fw_driver.update'
handler, ALSA PCM application can detect this situation by XRUN because
the target device cancelled to transmit packets once bus-reset occurs.
Due to the above mechanism, ALSA fireface driver just stops packet
streaming in the update handler, thus pcm.prepare handler should
request the reallocation.
This commit requests the reallocation in pcm.prepare callback when
bus generation is changed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Once allocated, isochronous resources are available for packet
streaming, even if the streaming is cancelled. For this reason,
current implementation handles allocation of the resources and
starting packet streaming at the same time. However, this brings
complicated procedure to start packet streaming.
This commit separates the allocation and starting. The allocation is
done in pcm.hw_params callback and available till pcm.hw_free callback.
Even if any XRUN occurs, pcm.prepare callback is done to restart
packet streaming without releasing/allocating the resources.
There are two points to stop packet streaming; in pcm.hw_params and
pcm.prepare callbacks. The former point is a case that packet streaming
is already started for any MIDI substream then packet streaming is
requested with different sampling transfer frequency for any PCM
substream. The latter point is cases of any XRUN or packet queueing
error.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a part of preparation to perform allocation/release
of isochronous resources in pcm.hw_params/hw_free callbacks.
This commit adds a helper function to allocate isochronous resources,
separated from operations to start packet streaming, I note that some
dice-based devices have two pair of endpoints for isochronous packet
straeming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a part of preparation to perform allocation/release
of isochronous resources in pcm.hw_params/hw_free callbacks.
There're three points to finish packet streaming but no helper
functions for common operations for it. This commit adds a helper
function for operations to finish packet streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Based on 1 normalized pattern(s):
licensed under the terms of the gnu general public license version 2
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-only
has been chosen to replace the boilerplate/reference in 88 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Alexios Zavras <alexios.zavras@intel.com>
Reviewed-by: Allison Randal <allison@lohutok.net>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190530000437.521539229@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Add SPDX license identifiers to all Make/Kconfig files which:
- Have no license information of any form
These files fall under the project license, GPL v2 only. The resulting SPDX
license identifier is:
GPL-2.0-only
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
