Mutex is doubly unlocked in some error path of pcm.open. This commit fixes
ALSA firewire-motu driver in Linux kernel v5.5.
Reported-by: kbuild test robot <lkp@intel.com>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Fixes: 3fd80b2003 ("ALSA: firewire-motu: use the same size of period for PCM substream in AMDTP streams")
Fixes: 0f5482e787 ("ALSA: firewire-motu: share PCM buffer size for both direction")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191208232226.6685-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loop optimizer seems to go astray here, and produces some warnings
that don't seem valid.
Still, the code can be simplified -- just clear the whole array at the
beginning, and fill in whatever values are valid on the platform.
Warnings before this change (GCC 8.2.0 ARM allmodconfig):
In file included from ../sound/pci/echoaudio/gina24.c:115:
../sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vumeters_get':
../sound/pci/echoaudio/echoaudio_dsp.c:647:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations]
In file included from ../sound/pci/echoaudio/layla24.c:112:
../sound/pci/echoaudio/echoaudio.c: In function 'snd_echo_vumeters_get':
../sound/pci/echoaudio/echoaudio_dsp.c:658:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations]
../sound/pci/echoaudio/echoaudio_dsp.c:647:9: warning: iteration 1073741824 invokes undefined behavior [-Waggressive-loop-optimizations]
Signed-off-by: Olof Johansson <olof@lixom.net>
Link: https://lore.kernel.org/r/20191207224953.25944-1-olof@lixom.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
syzkaller reported an invalid access in PCM OSS read, and this seems
to be an overflow of the internal buffer allocated for a plugin.
Since the rate plugin adjusts its transfer size dynamically, the
calculation for the chained plugin might be bigger than the given
buffer size in some extreme cases, which lead to such an buffer
overflow as caught by KASAN.
Fix it by limiting the max transfer size properly by checking against
the destination size in each plugin transfer callback.
Reported-by: syzbot+f153bde47a62e0b05f83@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191204144824.17801-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 609f548534 ("ALSA: hda: hdmi - preserve non-MST PCM
routing for Intel platforms") tried to restore the old behavior wrt
assignment of the PCM slot for Intel platforms, but this didn't do it
right. As found in the later discussion, a positive pipe id on Intel
platforms can be passed for single monitor attachment case.
This patch reverts the previous attempt and applies a simpler
workaround instead. Actually, for Intel platforms, we can handle as
if per_pin->dev_id=0, assign the primary slot at the first try. This
assures the compatible behavior with the previous versions regarding
the slot assignment.
Fixes: 609f548534 ("ALSA: hda: hdmi - preserve non-MST PCM routing for Intel platforms")
Link: https://lore.kernel.org/r/20191203154105.30414-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit in HD-audio stream management for changing the
stripe control seems causing a regression on some platforms. The
stripe control is currently used only by HDMI codec, and applying the
stripe mask unconditionally may lead to scratchy and static noises as
seen on some MacBooks.
For addressing the regression, this patch changes the stream
management code to apply the stripe mask conditionally only when the
codec driver requested.
Fixes: 9b6f7e7a29 ("ALSA: hda: program stripe bits for controller")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477
Tested-by: Michael Pobega <mpobega@neverware.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191202074947.1617-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-parser assigns the bass speaker to DAC3 (NID 0x06) which
is without the volume control. I do not see a reason to use DAC2,
because the shared output to all speakers produces the sufficient
and well balanced sound. The stereo support is enough for this
purpose (laptop).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191129144027.14765-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 5398e94fb7 ("ALSA: hda - Add DP-MST support for NVIDIA codecs")
introduced a slight change of behaviour how non-MST monitors are
assigned to PCMs on Intel platforms.
In the drm_audio_component.h interface, the third parameter
to pin_eld_notify() is pipe number. On Intel platforms, this value
is -1 for MST. On other platforms, a non-zero pipe id is used to
signal MST use.
This difference leads to some subtle differences in hdmi_find_pcm_slot()
with regards to how non-MST monitors are assigned to PCMs.
This patch restores the original behaviour on Intel platforms while
keeping the new allocation policy on other platforms.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20191129143756.23941-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add additional check in hdmi_find_pcm_slot() to not return
a pcm index that points to unallocated pcm. This could happen
if codec driver is set up in codec->mst_no_extra_pcms mode.
On some platforms, this leads to a kernel oops in snd_ctl_notify(),
called via update_eld().
BugLink: https://github.com/thesofproject/linux/issues/1536
Fixes: 5398e94fb7 ALSA: hda - Add DP-MST support for NVIDIA codecs
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20191129143756.23941-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've added the bass speaker support on Acer 8951G by the commit
00066e9733 ("Add Acer Aspire Ethos 8951G model quirk"), but it seems
that the GPIO pin was wrongly set: while the commit turns off the bit
to power up the amp, the actual hardware reacts other way round,
i.e. GPIO bit on = amp on.
So this patch fixes the bug, turning on the GPIO bit 0x02 as default.
Since turning on the GPIO bit can be more easily managed with
alc_setup_gpio() call, we simplify the quirk code by integrating the
GPIO setup into the existing alc662_fixup_aspire_ethos_hp() and
dropping the whole ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER quirk.
Fixes: 00066e9733 ("Add Acer Aspire Ethos 8951G model quirk")
Reported-and-tested-by: Sergey 'Jin' Bostandzhyan <jin@mediatomb.cc>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191128202630.6626-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
headphone have noise even the volume is very small.
Let it fill up pcbeep hidden register to default value.
The issue was gone.
Fixes: 4344aec84b ("ALSA: hda/realtek - New codec support for ALC256")
Fixes: 736f20a706 ("ALSA: hda/realtek - Add support for ALC236/ALC3204")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/9ae47f23a64d4e41a9c81e263cd8a250@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix regression in how intel_haswell_fixup_connect_list()
results are used in hda_read_pin_conn(). Use of
snd_hda_get_raw_connections() in hda_read_pin_conn() bypasses
the cache and thus also bypasses the overridden pin connection
list. On platforms that require the connection list fixup,
mux list will be empty and HDMI playback will fail to -EBUSY
at open.
Fix the regression in hda_read_pinn_conn(). Simplify code
as suggested by Takashi Iwai to remove old
intel_haswell_fixup_connect_list() and copy the cvt_nid list
directly and not use snd_hda_override_conn_list() at all.
Fixes: 9c32fea836 ("ALSA: hda - Add DP-MST support for non-acomp codecs")
BugLink: https://github.com/thesofproject/linux/issues/1537
Cc: Nikhil Mahale <nmahale@nvidia.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20191127161240.17026-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Static analysis tools (cppcheck and PVS Studio) report an error
in loopback_snd_timer_period_elapsed() regarding dpcm_play pointer
dereference earlier than its null-check. And although this is a result
of a formal check, and the pointer correctness is also protected
by having a corresponding bit set in the "running" mask, re-ordering
of the lines can imake the code even formally correct and eliminate
those static analysis error reports.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Reported-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191127110622.26105-1-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So that we can power down the GPU and audio to save power.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20191122214353.582899-5-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20191122214353.582899-4-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only enable the vga_switcheroo logic on systems with the
ATPX ACPI method. This logic is not needed for asics
that are not part of a PX (PowerXpress)/HG (Hybrid Graphics)
platform.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20191122214353.582899-2-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage.
This patch introduces this functionality as to be seen in the mac
or windows scarlett control.
The correct address could already be found in the dump info, but was
never used. Without this patch inputs 3/4 are quite unusable else.
Signed-off-by: Jens Verwiebe <info@jensverwiebe.de>
Link: https://lore.kernel.org/r/384d65cd-5e87-91eb-9fc3-e57226f534c6@jensverwiebe.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Laptops like ASUS UX431FLC and UX431FL can share the same audio quirks.
But UX431FLC needs one more step to enable the internal speaker: Pull
the GPIO from CODEC to initialize the AMP.
Fixes: 60083f9e94 ("ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FL")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191125093405.5702-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some more development work for v5.5. Highlights include:
- More cleanups from Morimoto-san.
- Trigger word detection for RT5677.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl3bzrQTHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0LkMB/4/MkpN4F2WudNCKgmYay2gw3hq8aOR
5Xh/SE6N3BEgBpQiMEUdePRT6LqnQMsSpOaUoKfd4nFvCyLaV7itaYzQuUpo+UJ/
a4UDHJPVfrDJeFghWqSngfEW/N66mpO3UuAvhSRTR3ku0T31v9FkcLsMbdxVUpLC
ablSaFfPCOtTQOsG9blRUL/GAWUhGzPI/Hl4VldS0FvKDmMDY2fxt7QxXRaEYItc
i1Uthklxd0BfRmDqv0WEpduUhbwzg6Uir9pXSisux7DpMk1Li0IyPST5UcRl0Zf4
CKL/qfdQANpufj9kGmrlCOnXB9P5/XYVwfYV430WqMyw0QaOzzSrxT7C
=CZ/+
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.5-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: More updates for v5.5
Some more development work for v5.5. Highlights include:
- More cleanups from Morimoto-san.
- Trigger word detection for RT5677.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An explicit Kconfig dependency is missing for the recent addition of
the timer support. CONFIG_SND_TIMER isn't always selected by SND_PCM.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Reported-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20191124083924.14049-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code:
LENOVO-20QE000VMC-ThinkPadX1Carbon7th-20QE000VMC
With the patch:
LENOVO-20QE000VMC-ThinkPadX1Carbon7th
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191120174435.30920-2-perex@perex.cz
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add append_dmi_string() function and make the code more readable.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191120174435.30920-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
The current driver only sets 0x76543210 and 0x67452301 for DALIGN.
This doesn’t work well for TDM split and ex-split mode for all SSIU.
This patch programs the DALIGN registers based on the SSIU number.
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Jiada Wang <jiada_wang@mentor.com>
Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Fixes: a914e44693 ("ASoC: rsnd: more clear rsnd_get_dalign() for DALIGN")
Signed-off-by: Nilkanth Ahirrao <anilkanth@jp.adit-jv.com>
Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20191121111023.10976-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
loopback_snd_timer_close_cable() function waits until all
scheduled tasklets are completed, but the timer is closed after that
and can generate more event callbacks, scheduling new tasklets,
that will not be synchronized with cable closing.
Move tasklet_kill() call to be executed after snd_timer_close()
call to avoid such case.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191122175218.17187-2-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
loopback_parse_timer_id() uses snd_card_ref(), that can lock on mutex,
also snd_timer_instance_new() uses non-atomic allocation, that can sleep.
So, both functions can not be called from loopback_snd_timer_open()
with cable->lock spinlock locked.
Moreover, most part of loopback_snd_timer_open() function body works
when the opposite stream of the same cable does not yet exist, and
the current stream is not yet completely open and can't be running,
so existing locking of loopback->cable_lock mutex is enough to protect
from conflicts with simultaneous opening or closing.
Locking of cable->lock spinlock is not needed in this case.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191122175218.17187-1-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of the new PCM sync_stop ops in ASoC component.
It's optional and can be NULL unless you need the sync operation.
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191121190709.29121-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now PCM core accepts the NULL ioctl ops as default, and passing a proper
ioctl ops is no longer mandatory. Adjust soc_new_pcm() to allow also
the NULL for component ioctl ops, too.
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191121190709.29121-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDMI codec may leave codec->relaxed_resume flag set even after
unbinding. Clear it unconditionally.
It's very unlikely that this actually matters in the real use case,
so just a fix for consistency.
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
Link: https://lore.kernel.org/r/20191122132624.5482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The old Nvidia chips have multiple HD-audio codecs on the same
HD-audio controller, and this doesn't work as expected with the current
audio component binding that is implemented under the one-codec-per-
controller assumption; at the probe time, the driver leads to several
kernel WARNING messages.
For the proper support, we may change the pin2port and port2pin to
traverse the codec list per the given pin number, but this needs more
development and testing.
As a quick workaround, instead, this patch drops the binding in the
audio side for these legacy chips since the audio component support in
nouveau graphics driver is still not merged (hence it's basically
unused).
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205625
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
Link: https://lore.kernel.org/r/20191122132000.4460-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return from pnp_irq is an unsigned integer type resource_size_t
and hence the error check for a positive non-error code is always
going to be true. A check for a non-failure return from pnp_irq
should in fact be for (resource_size_t)-1 rather than >= 0.
Addresses-Coverity: ("Unsigned compared against 0")
Fixes: a9824c868a ("[ALSA] Add CS4232 PnP BIOS support")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20191122131354.58042-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_mixer_controls_badd() that parses UAC3 BADD profiles misses a
NULL check for the given interfaces. When a malformed USB descriptor
is passed, this may lead to an Oops, as spotted by syzkaller.
Skip the iteration if the interface doesn't exist for avoiding the
crash.
Fixes: 17156f23e9 ("ALSA: usb: add UAC3 BADD profiles support")
Reported-by: syzbot+a36ab65c6653d7ccdd62@syzkaller.appspotmail.com
Suggested-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191122112840.24797-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s6i6_gen2_info.ports[] array had the Mixer and PCM port type
entries in the wrong place. Use designators to explicitly specify the
array elements being set.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191110134356.GA31589@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a new Dell machine which needs to apply the quirk
ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table
to fix it this time. And we could remove all pintbls of alc236
for applying DELL1_MIC_NO_PRESENCE on Dell machines.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20191121022644.8078-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a new Dell machine which needs to apply the quirk
ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table
to fix it this time. And we could remove all pintbls of alc256
for applying DELL1_MIC_NO_PRESENCE on Dell machines.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20191121022644.8078-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many PCI and other drivers performs snd_pcm_period_elapsed() simply in
its interrupt handler, so the sync_stop operation is just to call
synchronize_irq(). Instead of putting this call multiple times,
introduce the common card->sync_irq field. When this field is set,
PCM core performs synchronize_irq() for sync-stop operation. Each
driver just needs to copy its local IRQ number to card->sync_irq, and
that's all we need.
Link: https://lore.kernel.org/r/20191117085308.23915-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard programming model of a PCM sound driver is to process
snd_pcm_period_elapsed() from an interrupt handler. When a running
stream is stopped, PCM core calls the trigger-STOP PCM ops, sets the
stream state to SETUP, and moves on to the next step. This is
performed in an atomic manner -- this could be called from the interrupt
context, after all.
The problem is that, if the stream goes further and reaches to the
CLOSE state immediately, the stream might be still being processed in
snd_pcm_period_elapsed() in the interrupt context, and hits a NULL
dereference. Such a crash happens because of the atomic operation,
and we can't wait until the stream-stop finishes.
For addressing such a problem, this commit adds a new PCM ops,
sync_stop. This gets called at the appropriate places that need a
sync with the stream-stop, i.e. at hw_params, prepare and hw_free.
Some drivers already have a similar mechanism implemented locally, and
we'll refactor the code later.
Link: https://lore.kernel.org/r/20191117085308.23915-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently PCM ioctl ops is a mandatory field but almost all drivers
simply pass snd_pcm_lib_ioctl. For simplicity, allow to set NULL in
the field and call snd_pcm_lib_ioctl() as default.
Link: https://lore.kernel.org/r/20191117085308.23915-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the documentation for the newly introduced managed buffer
allocation mode accordingly. The old preallocation is no longer
recommended.
Link: https://lore.kernel.org/r/20191117085308.23915-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for the feature to automatically allocate
and free PCM buffers, so called "managed buffer allocation" mode.
It's set up via new PCM helpers, snd_pcm_set_managed_buffer() and
snd_pcm_set_managed_buffer_all(), both of which correspond to the
existing preallocator helpers, snd_pcm_lib_preallocate_pages() and
snd_pcm_lib_preallocate_pages_for_all(). When the new helper is used,
it not only performs the pre-allocation of buffers, but also it
manages to call snd_pcm_lib_malloc_pages() before the PCM hw_params
ops and snd_lib_pcm_free() after the PCM hw_free ops inside PCM core,
respectively. This allows drivers to drop the explicit calls of the
memory allocation / release functions, and it will be a good amount of
code reduction in the end of this patch series.
When the PCM substream is set to the managed buffer allocation mode,
the managed_buffer_alloc flag is set in the substream object. Since
some drivers want to know when a buffer is newly allocated or
re-allocated at hw_params callback (e.g. want to set up the additional
stuff for the given buffer only at allocation time), now PCM core
turns on buffer_changed flag when the buffer has changed.
The standard conversions to use the new API will be straightforward:
- Replace snd_pcm_lib_preallocate*() calls with the corresponding
snd_pcm_set_managed_buffer*(); the arguments should be unchanged
- Drop superfluous snd_pcm_lib_malloc() and snd_pcm_lib_free() calls;
the check of snd_pcm_lib_malloc() returns should be replaced with
the check of runtime->buffer_changed flag.
- If hw_params or hw_free becomes empty, drop them from PCM ops
Link: https://lore.kernel.org/r/20191117085308.23915-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show and change sound card timer source with read-write info
file in proc filesystem. Initial string can still be set as
module parameter.
The timer source string value can be changed at any time,
but it is latched by PCM substream open callback (the first one
for a particular cable). At this point it is actually used, that
is the string is parsed, and the timer is looked up and opened.
The timer source is set for a loopback card (the same as initial
setting by module parameter), but every cable uses the value,
current at the moment of open.
Setting the value to empty string switches the timer to jiffies.
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-8-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
to do synchronous audio forwarding between hardware sound card and aloop
devices. Such an audio route could look like the following:
Sound card -> Loopback application -> ALSA loop device -> arecord
In this case the loopback device should use the sound timer of the sound
card. Without this patch the loopback application has to implement an
adaptive sample rate converter to align the different clocks of the
different ALSA devices.
The used timer can be selected by referring to a sound card, its device
and subdevice, when loading the module:
$ modprobe snd_aloop enable=1 timer_source=[<card>[.<dev>[.<subdev>]]]
<card> is the name (id) of the sound card or a card number.
<dev> and <subdev> are device and subdevice numbers (defaults are 0).
Empty string as a value of timer_source= parameter enables previous
functionality (using jiffies timer).
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-7-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit does not change the behaviour. It only separates the jiffies
timer specific implementation from the generic part.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-5-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
