The newly added CODEC to CODEC DAI link widget pointers in
snd_soc_dai_link are better placed in snd_soc_pcm_runtime.
snd_soc_dai_link is really intended for static configuration of
the DAI, and the runtime for dynamic data. The snd_soc_dai_link
structures are not destroyed if the card is unbound. The widgets
are cleared up on unbind, however if the card is rebound as the
snd_soc_dai_link structures are reused these pointers will be left at
their old values, causing access to freed memory.
Fixes: 595571cca4 ("ASoC: dapm: Fix regression introducing multiple copies of DAI widgets")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200526161930.30759-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 54947cd64c ("ALSA: hda/realtek - Fix
speaker output regression on Thinkpad T570"). Essentially it fixes
the DAC / pin pairing by a static table. It was confirmed and merged
to stable kernel later.
Now, interestingly, we got another regression report for the very same
model (T570) about the similar problem, and the commit above was the
culprit. That is, by some reason, there are devices that prefer the
DAC1, and another device DAC2!
Unfortunately those have the same ID and we have no idea what can
differentiate, in this patch, a new fixup model "tpt470-dock-fix" is
provided, so that users with such a machine can apply it manually.
When model=tpt470-dock-fix option is passed to snd-hda-intel module,
it avoids the fixed DAC pairing and the DAC1 is assigned to the
speaker like the earlier versions.
Fixes: 54947cd64c ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570")
BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hi Mark
Current ALSA SoC is handling dai_link related operation,
but it is implmemented directly without using function/macro,
and at random place.
This v4 patch-set creates new snd_soc_link_xxx() functions
which handles dai_link related operation,
and implmement these at new soc-link.c.
v3 -> v4
- add Reviewed-by from Ranjani and Pierre-Louis
- fix bisection error at [2/7]
v2 -> v3
- add missing #include <sound/soc-link.h> in soc-link.c
v1 -> v2
- #include <sound/soc-link.h> is added on each c source file
instead of soc.h
- not have extra error message after snd_soc_link_xxx(),
because it already indicate it via snc_link_ret()
- snd_soc_link_compr_xxx() doesn't have rtd parameter,
because it can be created from cstream
Link: https://lore.kernel.org/r/87lflk4yk3.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/874ksa59wc.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/877dx868op.wl-kuninori.morimoto.gx@renesas.com
Kuninori Morimoto (7):
ASoC: add soc-link.c
ASoC: soc-link: move soc_rtd_xxx()
ASoC: soc-link: remove unneeded parameter from snd_soc_link_xxx()
ASoC: soc-link: add snd_soc_link_be_hw_params_fixup()
ASoC: soc-link: add snd_soc_link_compr_startup()
ASoC: soc-link: add snd_soc_link_compr_shutdown()
ASoC: soc-link: add snd_soc_link_compr_set_params()
include/sound/soc-link.h | 27 +++++++
sound/soc/Makefile | 2 +-
sound/soc/soc-compress.c | 46 ++++--------
sound/soc/soc-core.c | 18 ++---
sound/soc/soc-dai.c | 9 ++-
sound/soc/soc-link.c | 150 +++++++++++++++++++++++++++++++++++++++
sound/soc/soc-pcm.c | 86 ++++------------------
7 files changed, 219 insertions(+), 119 deletions(-)
create mode 100644 include/sound/soc-link.h
create mode 100644 sound/soc/soc-link.c
--
2.17.1
Thank you for your help !!
Best regards
---
Kuninori Morimoto
Hello,
extended firmware manifest is a method to retrieve capabilities
directly from the firmware file instead of routing the information via
the DSP and reading it back via IPC (latter mechanism still supported
but will be deprecated).
This feature was briefly merged to 5.8 with the series sent on
2020-Apr-15, but due to a regression hit with exporting uapi headers,
the patches got dropped.
Here's an update with the uapi header issue fixed, rebased to latest
'for-5.8' and a few minor fixes. This has been sitting in sof-dev for
some weeks and no further issues have been found. We also added
a check for the uapi-export case to SOF CI, so such errors would
not slip through again in the future.
Tooling support to create firmware files with an extended header
is available in SOF firmware repository (see the rimage tool) and
this part is already merged.
Karol Trzcinski (5):
ASoC: SOF: loader: Adjust validation condition for fw_offset
ASoC: SOF: Introduce extended manifest
ASoC: SOF: ext_manifest: parse firmware version
ASoC: SOF: ext_manifest: parse windows
ASoC: SOF: ext_manifest: parse compiler version
include/sound/sof/ext_manifest.h | 95 +++++++++++++++++
sound/soc/sof/intel/hda-loader.c | 2 +-
sound/soc/sof/loader.c | 176 ++++++++++++++++++++++++++++++-
3 files changed, 269 insertions(+), 4 deletions(-)
create mode 100644 include/sound/sof/ext_manifest.h
--
2.26.2
COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=x86_64
sound/soc/fsl/fsl_asrc.c:557:18: warning: no previous prototype for function 'fsl_asrc_get_dma_channel' [-Wmissing-prototypes]
struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir)
^
sound/soc/fsl/fsl_asrc.c:557:1: note: declare 'static' if the function is not intended to be used outside of this translation unit
struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir)
^
static
Fixes: be7bd03f02 ("ASoC: fsl_asrc: Move common definition to fsl_asrc_common")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: shengjiu wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1590397412-12966-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Updating power management register requires toggling the shutdown
bit otherwise it causes unexpected behavior of the codec.
However, toggling the shutdown bit results in loud speaker crackling.
Setup the power management register only at startup to avoid that.
Signed-off-by: Pavel Dobias <dobias@2n.cz>
Link: https://lore.kernel.org/r/20200522142957.18364-3-dobias@2n.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound fixes from Takashi Iwai:
"Just a few small fixes: the only significant one is a slight
improvement for PCM running position update with no-period-elapsed
case while the rest are HD-audio fixups and ice1712 model quirk"
* tag 'sound-5.7-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Add more fixup entries for Clevo machines
ALSA: iec1712: Initialize STDSP24 properly when using the model=staudio option
ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme
ALSA: pcm: fix incorrect hw_base increase
In MOTU protocol, data block consists of SPH and 24-bit chunks
aligned to quadlet. The number of chunks per data block is specific
to model. For models with optical interface, the number differs
depending on I/O settings for the interface (ADAT, TOSLINK).
Currently the number is calculated from flags in model-specific
data. However this is weak in the case that the model has quirks.
Actually, for quirks of some models, flags are used against their
original meanings.
This commit adds model-specific table of chunk count. For future
integration, this table is based on the calculation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200519111641.123211-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ranjani Sridharan