Update RX shutdown sequence so that codec
path gets tear down first followed by cpu dai.
This will avoid slim port underflow/overflows
when slim data protocol is changed.
Change-Id: I6e3582fa010d18d4e0ccfde319dfc4d81af1351f
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Fix the soc_find_component function to make sure either the of_node or
the name is provided to compare against the registered components to fix
possible NULL pointer de-reference.
CRs-fixed: 925138
Change-Id: Ic1f02c341c06cadcfe6de638ff6c86e51845e59f
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Channel count for SLIM_RX_6 port cannot be set
since get_port_idx() returns invalid port id which
resulting in invalid channel configuration for headset
usecase. Fix by adding SLIM_RX_6 case in get_port_idx().
Change-Id: Iadd3e995d044198c711f744c11b62cec2f7902c0
Signed-off-by: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
Slimbus TX 7 and 8 would need to be connected to Slimbus RX 6
for different use cases using loopback in AFE. Updated necessary
routings for supporting the loopback.
CRs-Fixed: 1036018
Change-Id: I46c797a6550884bf42a2d7763590047d2e750906
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Size of param buffer should be big enough to hold param length
of data and param payload.
CRs-Fixed: 1033525
Change-Id: I6fa58f87a7c7df5f0485ea5b368ea090eb8bedb4
Signed-off-by: Ashish Jain <ashishj@codeaurora.org>
Backend DAIs are not enabled for low-latency-record
bt-sco. Update mixer control array of MultiMedia5
mixer to enable backend DAIs.
CRs-Fixed: 1029460
Change-Id: I8e01302baf2d78afca930ef1f251906a971a8234
Signed-off-by: Karthik Reddy Katta <a_katta@codeaurora.org>
During non-gapless transition, there is an indefinite wait in
drain until either eos_ack or cmd_interrupt is set. This results
in playback getting stuck and occurs because cmd_interrupt is
not set in TRIGGER_STOP as gapless_transition is set to 1 during
partial drain of earlier stream.
Fix the issue by setting gapless_transition to 0 when gapless
fails which ensures that cmd_interrupt is set in TRIGGER_STOP.
CRs-Fixed: 1027991
Change-Id: I47d2d45df8686f25e8170a84fcaf68e143f6e4f6
Signed-off-by: Satya Krishna Pindiproli <satyak@codeaurora.org>
Changes to support packed 24 bit (SNDRV_PCM_FORMAT_S24_3LE).
CRs-Fixed: 1011048
Change-Id: I5c49091d6bbff98ed8665446fffdba08446073cd
Signed-off-by: Manish Dewangan <manish@codeaurora.org>
Changes to support packed 24 bit (SNDRV_PCM_FORMAT_S24_3LE).
CRs-Fixed: 1011048
Change-Id: If81f3053629dc4f80a08392f392c7be735ad33c2
Signed-off-by: Manish Dewangan <manish@codeaurora.org>
Driver changes to use ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 command.
This command supports playback/record of both 32 bit
(24 bit data in 32 bit word) and 24 bit packed. Update platform
drivers to use this for SNDRV_PCM_FORMAT_S24_LE record and playback.
CRs-Fixed: 1011048
Change-Id: I6f98bf3402a737bc21daff33b13b137850a690ea
Signed-off-by: Manish Dewangan <manish@codeaurora.org>
With single voice architecture, two new VSIDs are created to
support multimode voice call. Update host pcm driver to support
new VSIDs.
Change-Id: I42e33db7f3dca47c30b7dc5af59848eb6beef330
Signed-off-by: Helen Zeng <xiaoyunz@codeaurora.org>
Validate input data length to ensure only relevant data
is copied.
CRs-Fixed: 1027585
Change-Id: I67eb4f162f944bbf4d9e55fb8fe93759e6b8ff91
Signed-off-by: Ashish Jain <ashishj@codeaurora.org>
Timeout error is observed while waiting for
ADM_CMD_SET_PP_PARAMS_V5 command's response.
Fix the condition logic in wait_event_timeout()
to match the value set in adm_callback() when
response to ADM_CMD_SET_PP_PARAMS_V5 is received.
CRs-Fixed: 1030674
Change-Id: I711c860dc3de479eec0d22369d19615aef572ea1
Signed-off-by: Karthik Reddy Katta <a_katta@codeaurora.org>
Set the stream open flag immediately after the
stream is opened to ensure correct closure of
the stream if there is an error condition.
Change-Id: I61faf6ddf99ab504e492a4e37d577b67acf99f09
Signed-off-by: Ben Romberger <bromberg@codeaurora.org>
Enable use of noirq (i.e pull mode and push mode)
playback and capture.
CRs-Fixed: 992798
Change-Id: I98e68c2a485783be3c2b3eaa62577759d7e21d82
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Add capture support to MultiMedia3 frontend.
CRs-Fixed: 992798
Change-Id: Ie21a1c4a73c354a6dc1e733e6d2ac653f85f7647
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Implement platform drivers to support shared memory based
pcm playback and capture.
Change-Id: I882c67ae1c3d950b98bd002ac384cc3a7e77874a
CRs-Fixed: 992798
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
Sending PP params and calibration params for compress
passthrough path is resulting in timeout which is
delaying the start of playback.
Sending the PP params only when it is legacy pcm playback.
Change-Id: I7fe2840b7a72bddde887340a6e913cb120d1bc61
CRs-Fixed: 1030688
Signed-off-by: Satish Babu Patakokila <sbpata@codeaurora.org>
Currently ASoC core creates a static route b/w
playback/capture widgets of cpu and codec dai
if they are part of the same dai-link. However
this will cause codec path to get powered up first
followed by the backend dai start during device
switch use-case where the front-end is not closed,
leading to audio playback failure if either bit-width
or sample rate is different.
CRs-Fixed: 1029118
Change-Id: I180515f2ad55d1f446ad7eb1ad0bd71809db94bd
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Add pointer validation checks to prevent sending
invalid handles to ADSP as part of unmap memory
regions command.
CRs-Fixed: 1018367
Change-Id: I0dfb2fccb4414ed82ee10d73576fda66a273043d
Signed-off-by: Karthik Reddy Katta <a_katta@codeaurora.org>
When routing DMIC input to ANC block for handset ANC usecase,
codec driver enters an infinite loop attempting to determine
the stream sample rate. Additionally since the noise DMIC is
configured prior to the rest of the usecase, we cannot deterine
the stream sample rate to configure the ANC block for half-rate.
Therefore revert that logic and let ANC block be configured
according to the device tree.
CRs-fixed: 997662
Change-Id: I311ad8f158b0be6e9d6481512860f9fac10afc1f
Signed-off-by: Stephen Oglesby <soglesby@codeaurora.org>
Currently DMIC clock is set at 4.8MHz for all sampling rates. For
optimal power, sampling rates <=48KHz should be set to 2.4MHz.
CRs-fixed: 971183
Change-Id: If3076f017d476cfb57fa22b75cc74ed615c8882e
Signed-off-by: Stephen Oglesby <soglesby@codeaurora.org>
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
cmd_interrupt flag is set during first stream's stop in gapless playback
but it is not reset after receiving eos ack. This interrupts second
stream partial drain and eos is sent to client, which leads to session
close causing audio mute. Do not set cmd_interrupt during gapless
transition to fix the issue as no one is waiting for eos.
CRs-Fixed: 1012546
Change-Id: Ibcbdde0ea59ff80a798de0b894c2239899260860
Signed-off-by: Dhanalakshmi Siddani <dsiddani@codeaurora.org>
Moisture detection is needed only for NO jack type.
So disable moisture detection feature for NC Jack.
CRs-Fixed: 1012001
Change-Id: I93f72f18145ddef6a0caf2c59a9af5f23e6e20a3
Signed-off-by: Yeleswarapu Nagaradhesh <nagaradh@codeaurora.org>
Switch to swap ground and mic headset poles is controlled by a
GPIO on the Apps processor instead of the PMIC, and therefore
software logic must change to use pinctrl APIs
CRs-fixed: 1019254
Change-Id: Ibccddc82b18614ddbe6ef9c9720b3de1ce00163e
Signed-off-by: Stephen Oglesby <soglesby@codeaurora.org>
In wcd9330 driver, external clk enable callback function
is passed with argument as true always, instead of passing
the arguments from caller. This is leading to mclk users
count to increase without check.
CRs-fixed: 1013573
Change-Id: I113657c91dd5eb00791535dc78b7cdad1db5c4aa
Signed-off-by: Viraja Kommaraju <virajak@codeaurora.org>
If long button is pressed to end the voice call, the button
click suppression block within wcd9335 hardware does not
release IN2_P causing TX mute for the next voice call session.
Avoid TX mute by force release IN2_P during every voice call
start.
CRs-fixed: 1013280
Change-Id: I5af41bef6db6af14d53018caef1f7fd9b00fc136
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Add EC reference support for USB audio ADSP solution so that
the USB audio rx can be used for echo cancellation.
Change-Id: If99081c1fd356e69710c94441affec92fac24075
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
Enable HDMI RX for 8996, otherwise soundcard
will not get registered for the flavors which
supports HDMI.
CRs-Fixed: 1023892
Change-Id: I0d2442c7b3d156ad919626a6015f0fbbf2116c3f
Signed-off-by: Yeleswarapu Nagaradhesh <nagaradh@codeaurora.org>
Parameter fragments and fragment_size are type of u32. U32_MAX is
the correct check.
CRs-Fixed: 1014726
Change-Id: Ia6d4755408646ac4a75724f3c6f2177651875da3
Signed-off-by: Xiaojun Sang <xsang@codeaurora.org>
Currently the AFE input port is connected to LSM while sending operation
mode parameter to CPE. It is possible that in certain cases, the operation
mode does not need to be sent at all. In such case, the input port still
needs to be connected. Fix this by moving the connection to AFE input port
during LSM_START so everytime LSM is started, it is connected to the
correct AFE port.
CRs-fixed: 1012715
Change-Id: I6dbc344d5d7063c7cfd2fb29c2c39fdee1250bbf
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
Change all qdsp6v2 audio driver to use %pK instead
of %p. %pK hides addresses when the users doesn't
have kernel permissions. If address information
is needed echo 0 > /proc/sys/kernel/kptr_restrict.
Change-Id: I7baa9f127266726fecf9238167a1e0128a258847
Signed-off-by: Ben Romberger <bromberg@codeaurora.org>
Request device ungroup of speaker channels for independent
disable. It is possible that stereo speaker channels can be
disabled one after other, so remove them from group otherwise
speaker can be left in enabled state.
CRs-fixed: 1007465
Change-Id: I358ab4edcb85ec65b064ca28368ad744f2d36870
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Add support for 48x2 frame structure in soundwire
so that when slave device data path is not enabled,
all control messaging will happen with 48x2 frame.
Soundwire slave devices send an explicit request to
enable data path which in turn change the frame
structure to 48x16.
CRs-fixed: 996586
Change-Id: Ia4329ac982eb2a29a2b925897cd87ca9711c30e3
Signed-off-by: Phani Kumar Uppalapati <phaniu@codeaurora.org>
Add slimbus 6 playback hostless and slimbus_6_rx back-end
dai-link to enable independent backend for different devices
during audio playback.
Change-Id: Idac26ac45f1177db96fc3fb5d4a5e2f837f86d1b
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
Add USB audio via ADSP support in the machine driver.
Change-Id: I9773555fb025a41afd27e078f6ef23a4d140128f
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
commit addacd801e1638f41d659cb53b9b73fc14322cb1 upstream.
The HD-audio reconfig function got broken in the recent kernels,
typically resulting in a failure like:
snd_hda_intel 0000:00:1b.0: control 3:0:0:Playback Channel Map:0 is already present
This is because of the code restructuring to move the PCM and control
instantiation into the codec drive probe, by the commit [bcd96557bd:
ALSA: hda - Build PCMs and controls at codec driver probe]. Although
the commit above removed the calls of snd_hda_codec_build_pcms() and
*_build_controls() at the controller driver probe, the similar calls
in the reconfig were still left forgotten. This caused the
conflicting and duplicated PCMs and controls.
The fix is trivial: just remove these superfluous calls from
reconfig_codec().
Fixes: bcd96557bd ('ALSA: hda - Build PCMs and controls at codec driver probe')
Reported-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2da2dc9ead232f25601404335cca13c0f722d41b upstream.
For reducing the noise from the headset output on ASUS UX501VW,
call the existing fixup, alc_fixup_headset_mode_alc668(), additionally.
Thread: https://bbs.archlinux.org/viewtopic.php?id=209554
Signed-off-by: Kaho Ng <ngkaho1234@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 84add303ef950b8d85f54bc2248c2bc73467c329 upstream.
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2d2c038a9999f423e820d89db2b5d7774b67ba49 upstream.
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug. It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Fixup callback is added for dais which
do not follow the FE and BE convention
and is directly controlled by userspace
such as hostless dais. This will restrict
the hw_params based on what is supported by
hardware rather than blindly setting what
is given by userspace.
Change-Id: I401c70ab5de1df10363ec808cb68f72d8d74af96
Signed-off-by: Anish Kumar <kanish@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
While setting up route for a particular device, compare
stream name of CPU DAI and Backend DAI to find the correct
Backend DAI.
Change-Id: Ic3f7c0e5b2a1055e7fdf52c78ded797a9a126d03
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Add new USB rx and tx afe ports and routing to different
fe dais to enable USB audio via ADSP.
Change-Id: I4f82ba27becee1f3b62c410be0d00876961f9b18
Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
Signed-off-by: Kuirong Wang <kuirongw@codeaurora.org>
Debugfs directory creation failure are not critical error.
However, the failure messages might be misleading and might
be interpreted as geniune failure in ASoC functionality.
Mark the failure messages as debug level.
Change-Id: Id61c81753d493b6508cbe87c59077adda4675ada
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Add machine driver code to support audio on MSMCOBALT based
boards with WCN3990 BT/FM chipset.
Change-Id: Ia23572f44775a04c8f8c67e9a61d6b9be8869b82
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
