"LA.UM.9.14.1.r1-14100-QCM6490.QSSI14.0"
* tag 'LA.UM.9.14.1.r1-14100-QCM6490.QSSI14.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc: Fix out-of-bound write
dsp: q6voice: Adds checks for an integer overflow
dsp: q6voice: Adds checks for an integer overflow
Change-Id: Ieb4b3e8ead0616624b7492cfc42bb72a024525b7
"LA.UM.9.14.r1-24700-LAHAINA.QSSI15.0"
* tag 'LA.UM.9.14.r1-24700-LAHAINA.QSSI15.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
Asoc: dsp: Fix to check the list is empty or not
dsp: q6lsm: Check size of payload before access
Fix for OOB access issue
dsp: q6asm increase the locking range
ASoC: msm-pcm-q6-v2: Add size check
ASoC : Add proper copyright marking.
dsp: afe: Add check for num_spks
soc: pinctrl-lpi: remove pm ops
asoc: msm-compress: Fix compress_pause failure on gki
Audio legacy: Integer overflow in msm_lsm_ioctl_compat during audio playback usecase. size = sizeof(p_info_32) + p_info_32.param_size; This overflow issue may result heap overflow during copying the data: memcpy(param_info_rsp, &p_info_32, sizeof(p_info_32));
ASoC : Add macro to differentiate auto code
ASoC: msm: get CoPP index based on FE id
asoc: codecs: add array bound check
soc: Address SWR rate mismatch interrupt
asoc : add configuration about SLIMBUS_7_TX
asoc: lahaina: add support for aud_ref_clk_sel mux
dsp: afe: add support for aud_ref_clk_sel mux
ASoC: dsp: Release lock before return
ASoC: wcd937x: Add flag to decide RX_MUTE for HPHL and EAR
ASoC: bolero: Add check for CMPDR switch
ASoC : add support of HDMI controller for rb3gen2 platform
asoc: lahaina: fractional sample rate support for TDM
asoc: support for fractional sample rate over tdm
Revert "asoc: msm-compress : Fix for CTS-on-gsi with gki"
audio-kernel:swr: Add dynamic SWRM clk support
asoc: codec: make mclk freq configurable in Bolero
asoc: lahaina: Add ext clk source support
asoc: ext-clk: Add support for configuring ext clk
dsp: afe: Enhance and expose API for configuring ext clk
dsp: q6voice: Handle mutex lock-unlock
asoc: msm-compress : Fix for CTS-on-gsi with gki
Change-Id: I21b8e5f5ef608bf298230419123e4ee37c9b0797
strlen(swr_dmic_name_prefix_of) + 1 bytes are allocated for prefix_name just above
techpack/audio/asoc/codecs/swr-dmic.c:685:11: error: size argument in 'strlcpy' call appears to be size of the source; expected the size of the destination [-Werror,-Wstrlcpy-strlcat-size]
strlen(swr_dmic_name_prefix_of) + 1);
~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~
1 error generated.
Change-Id: I808b4f135d42fc50587eeb9348848f7e0a3561cc
Signed-off-by: Alexander Winkowski <dereference23@outlook.com>
Checkpatch tool generates some errors and warnings due to added PM QoS feature
This change fixes these issues.
Change-Id: I0843ecd33144d7ac65649ee90ed156e97a1a2f8a
Signed-off-by: George Gao <georgao@codeaurora.org>
They're used only in their own source files so just mark them static.
Change-Id: Ie8699f97074a5838af4bc6eab4c1108258b97e4c
Signed-off-by: Alexander Winkowski <dereference23@outlook.com>
Cross-checked with msm-4.14 and msm-5.15 tags. The change does not exist
as a separate commit and is squashed into the initial import.
Change-Id: Ib3defedf0fbd5b7cdccdfba26383313ff05b4ef0
Signed-off-by: Alexander Winkowski <dereference23@outlook.com>
"LA.UM.9.14.r1-23100-LAHAINA.QSSI14.0"
* tag 'LA.UM.9.14.r1-23100-LAHAINA.QSSI14.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
Revert "asoc: msm-compress : Fix for CTS-on-gsi with gki"
asoc: msm-compress : Fix for CTS-on-gsi with gki
ASoC: msm-pcm-host-voice: Handle mutex lock-unlock
Change-Id: I84217298ebd04c41daa978fcd3321a3c1bd2670b
"LA.UM.9.14.r1-22400-LAHAINA.QSSI13.0"
* tag 'LA.UM.9.14.r1-22400-LAHAINA.QSSI13.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
Asoc: add functionality to get source Track data from FNN module
dsp: add support to get Source Track data from FNN module
asoc: lahaina: add 16 channel support on tdm interface
asoc: Add jitter cleaner external clock switch
dsp: q6asm: Add check for ADSP payload size
ASoC: msm-pcm-host-voice: Check validity of session idx
dsp: Address information leak in dsp drivers
dsp: q6lsm: Address use after free for mmap handle
dsp: q6afe: Add check for fbsp state
dsp: q6lsm: Add check for payload buffer
ASoC: Resolve use after free in listen sound client
ASoC: dsp: q6core: Avoid use after free
ASoC: msm-lsm-client: Integer overflow check
asoc:msm-lsm-client: check for param size before copying
Change-Id: Ia01bfa4467f137efa4578708177862b794aeaf91
"LA.UM.9.14.1.r1-10900-QCM6490.QSSI14.0"
* tag 'LA.UM.9.14.1.r1-10900-QCM6490.QSSI14.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
dsp: q6lsm: Address use after free for mmap handle
ASoC: Resolve use after free in listen sound client
dsp: q6lsm: Add check for payload buffer
ASoC: dsp: q6core: Avoid use after free
dsp: afe: check for param size before copying
dsp: q6core: validate payload size before access for AVCS
asoc: copyright fix
Change-Id: I6e86a207e0a4651c87724cc8333258bd1d7b5dce
"LA.UM.9.14.r1-22200-LAHAINA.QSSI14.0"
* tag 'LA.UM.9.14.r1-22200-LAHAINA.QSSI14.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
ASoC: msm-pcm-q6-v2: Add dsp buf check
ASoC: Add the judgment for TDM
asoc: swrm: disable bus reset based on swrm verison
dsp: afe: Add check for sidetone iir config copy size
ASoC: msm-pcm-host-voice: Address buffer overflow in hpcm playback copy
ASoC: msm-pcm-voip: Avoid interger underflow
dsp: asm: validate payload size before access
dsp: afe: check for param size before copying
dsp: q6core: validate payload size before access for AVCS
dsp: afe: Add check for num_channels
soc: swr-mstr-ctrl: add new lock to sync runtime_resume and runtime_suspend
soc: reduce the auto suspend timeout when swr event finished
Change-Id: I82d3446a3ecf6cf47f75ab34a292557f24339ebb
The reg in soc_dapm_mux is 32-bit. The BE DAI ID passed
as shift(to be operated on the reg) may be more than 31,
which may cause overflow.
Set reg field to SND_SOC_NOPM to avoid any DAPM operation
while passing BE IDs in shift_l field and hence avoid overflow.
Change-Id: Ibbbca04c61b7c56eb4c5a7485a4e93dc28a09709
Signed-off-by: Soumya Managoli <smanag@codeaurora.org>
[dereference23: Forward port to msm-5.4]
Signed-off-by: Alexander Winkowski <dereference23@outlook.com>
This causes parts of the audio module to be rebuilt during every
incremental build, even if there are no changes:
CC techpack/audio/ipc/apr.o - due to command line change
CC techpack/audio/ipc/apr_v2.o - due to command line change
CC techpack/audio/ipc/apr_tal_rpmsg.o - due to command line change
CC techpack/audio/ipc/wcd-dsp-glink.o - due to command line change
We're only experiencing this issue in techpack/audio/ipc at the moment,
but kill the timestamp injection in all the audio components to
eliminate the possibility of encountering this issue again in the
future. This is harmless since the injected BUILD_TIMESTAMP macro is
never used.
Change-Id: I35b588a2a902438b2c79ebbac31453f2e528bbbe
Signed-off-by: Danny Lin <danny@kdrag0n.dev>
"LA.UM.9.14.r1-21300-LAHAINA.QSSI12.0"
* tag 'LA.UM.9.14.r1-21300-LAHAINA.QSSI12.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc: Fix possible overflow in routing driver
asoc : Update proper comment for endpoint 2
Change-Id: I14f8b619785911a1ac2500fad64eccd290af1fff
"LA.UM.9.14.r1-21200-LAHAINA.QSSI13.0"
* tag 'LA.UM.9.14.r1-21200-LAHAINA.QSSI13.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc : mark used cal block as stale
asoc: Add build option for qcs6490
asoc: kona: Add QUAD I2S EBIT clk define to kona
ASoC: Add check for pm_qos_req before removal
ASoC: msm-pcm-q6-v2: Add dsp buf check
dsp : assign null to pointer that has freed address
asoc : send chanel mixer while rerouting sound device
asoc: lahaina: set correct swr port map when wcd disabled
asoc: wcd938x/wcd937x: add null pointer check for mbhc variable
asoc: wsa881x: Fix to set the correct volume level
asoc : enable mclk after ssr for talos
ASoC: Add support "adm cmd register" and "adm pp" events
dsp: Remove lock in voice_map_cal_memory
Change-Id: Id91693ecd7a7f7b85f1e96f1337ad4fba3b0d2b4
"LA.UM.9.14.r1-21000-LAHAINA.QSSI13.0"
* tag 'LA.UM.9.14.r1-21000-LAHAINA.QSSI13.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc: Update wcd937x codec driver controls
uapi:align msm_audio_ar.h with AR
asoc : modify spf_machine_dlkm module name for GVM
asoc: tavil: modify TX routes and dapm widgets
asoc: tavil: modify RX routes and dapm widgets
asoc: Fix App Type Gain can`t work proper
asoc: destroy mutex during module_exit
asoc: add return value check for channel mapping function
Asoc: 24bit support for output devices
Change-Id: Ic6f52c296b15b5e202da6c98668965d8ae997d6f
"LA.UM.9.14.r1-20700-LAHAINA.QSSI13.0"
* tag 'LA.UM.9.14.r1-20700-LAHAINA.QSSI13.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc: use virt_sndcard_ctl node for sndcard status update
dsp : search cal block using path in asm
asoc: audio reach kernel bringup on 8255 GVM
ASoC: codecs: Deinitialize mbhc on codec remove
asoc: Add kcontrol "QUAT_TDM_RX_7 Port Mixer TERT_TDM_TX_7"
audio: Modified SSR handling after sub-system shutdown compltes
asoc: remove error from function msm_pinctrl_mclk_enable
Change-Id: I3bad495ed1fe4ec8a3f7c2acb515c9acd96f31da
"LA.UM.9.14.r1-20500.01-LAHAINA.QSSI12.0"
* tag 'LA.UM.9.14.r1-20500.01-LAHAINA.QSSI12.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
asoc: add more return when msm_audio_ion_alloc failed
audio-kernel: Do SRC with MFC
dsp: dyn mclk support enable/disable in afe_set_lpass_clock_v2
asoc: ignore afe resp status to avoid 2s timeout
asoc: enables paired rx port to address tdm slot shift issue
dsp: functions for paired rx port enable/disable usecases
asoc: change HS_IF dummy to playback
audio-kernel: Reset client info on deregister.
audio-routing: Support Rx to Tx loopback
asoc: fixing incorrect log messages
kernel: add place_marker for sound card register
asoc: SEC CLK enable
ASoC : Add new kcontrol for starting ASRC
soc: swr-mstr-ctrl: Fix swr overflow for wsa
audio-kernel: Disable clock voting logs
asoc: 24 and 32bit format support for ASM loopback and playback
config: remove unused conf for arauto
audio-kernal: Loading only Gen3 specific mixer controls for Gen3 boards
kernel: add place_marker for sound card register
kernel: Add notifier for SSR
asoc: kona: check if codec is enabled
asoc: kona: Add wsa8815 codec and wsa883x codec init
Asoc: check for invalid voice session id
asoc: update port config for kona.
config: Enable digital codec resource manager config
asoc: auto: send effect config based on apptype
asoc: Update dai link for wsa codec dai.
asoc: codecs: Add nullptr check
Audio-kernel: Comparison of unsigned value
asoc : AFE Limiter module control
soc: wsa881x: move analog init complete flag
Change-Id: I566e3efbcd6a900e303bb89fb116e73c8de08bcf
"LA.UM.9.14.r1-19600.01-LAHAINA.QSSI12.0"
* tag 'LA.UM.9.14.r1-19600.01-LAHAINA.QSSI12.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
kernel: Check for buffer overflow and NULL pointer
asoc: Add LLNP mode support
asoc: Update machine driver for kona
ASoC: codecs: Force enable micbias standalone
audio-kernel: add ar conf in makefie
ipc: apr: Add config to control the ipc context print
asoc: Disable chmixer override flag
asoc: get and put for SEC_TDM_RX_0/TX_0 ADM channels
asoc: msm-pcm-routing: Assign default topology when app_type is zero
ASoC: wcd937x: Add check for ULP irrespective of the Class
asoc: wsa881x: Fix panic on stress reboot
asoc: sm6150: change msm_wcn_be_dai_links name
Change-Id: Ia587e1021992c7e6d43710fbb6e20c60fe7be100
"LA.UM.9.14.r1-19500-LAHAINA.QSSI12.0"
* tag 'LA.UM.9.14.r1-19500-LAHAINA.QSSI12.0' of https://git.codelinaro.org/clo/la/platform/vendor/opensource/audio-kernel:
ASoC: optimize automotive routing drivers
ASOC: automotive routing drivers base version
Revert "ASOC: Disable Temperature sensor of WSA881x for bengal"
asoc: add afe limiter disable kcontrol and fix ULLPP index mismatch
asoc: codec: Fix error makefile: Fix error and add version information
ASOC: talos: update dai-link port-config for talos
ASoC: wcd937x: Update QCRG sequence for wcd937x
dsp: lower the log lever of ADSP_EALREADY
dsp: Remove error log when there is no AFE cal data
asoc: reduce HS TDM widget connection when it's disabled
soc: remove tdm slot mask invalid err log in multi lane case
asoc: Add VI feedback dai_link for bengal
asoc: msm: Add VI feedback mixer
asoc: wsa881x: Add mixer for speaker protection
ASOC: update sm6150 machine driver
asoc: wcd937x: update mic bias registers for new fab
ASOC: talos: update dai-link port-config for talos
asoc: codec: Fix CFI error caused by missing members
ASOC: update wcd934x codec driver
ASOC: update sm6150 machine driver
ASoC: wcd-spi: Fix dma alloc compilation errors
makefile: add config support for audio techpack for QCS610
ASoC: codecs: wsa881x: add codec dai for each wsa881x
soc: swr-mstr-ctrl: Fix SWR cmd FIFO overflow error
ASOC: ext-clk: Fix hw init issue caused by CLK upgrade
asoc: Set clock index correctly in mi2s slave mode
asoc:msm-lsm-client: potential deadlock fix
ASoC: msm: support multi group matrix mixer configure for SSMD mode
msm-audio: adding mixer control for soft step volume params
dsp: support ADM output map for less then 2 channels
asoc: Enable all LPAIF interface
asoc: afe: Add afe dyn mclk kcontrol
ASOC: Disable Temperature sensor of WSA881x for bengal
asoc: refine dummy machine driver for gen3
ASoC: msm: extend app type cfg mixer command with copp mode
ASoC: bitra: Add support for VI sense backend
asoc: copyright fix
ASoC : add support for MultiMedia17
dsp: Update adsp packet timeout in voice drv
ASoC: Unmap lab buffer during listen session close
audio-kernel: change the maxnum for stream in QACT
asoc: set default value for port_id of logging kcontrol
ASoc: bolero: update logic to disable VA_MCLK
ASoC: bolero: Disable va_swr gpio on clk failure
dsp: add new boot cmd for early audio
ARM: dts: msm: Change default state to sleep
audio-kernel: lvgvm 8295 related file modify
asoc: bengal: fix dailink init error
asoc: msm_stub: increase the max ch to 32
asoc: QUAT TDM support for dummy machine driver
dsp: Make synchronous call to unmap afe cal data
dsp:q6afe: Fix avcs load module failure for aptx
ASoC: msm: Fix ASRC port type mismatch
soc: add snd_event_init func for mark audio KPI
ASoC: wcd937x: Update EAR PA CnP FSM sequence
asoc: fix error for VA_CDC_DMA_TX_0 for bengal
asoc: codec: enable use_single_read for wsa and wcd regmap
config: sa8155/sa6155/gvmq: remove unused config vars
asoc: fix compile warning of format-extra-args
soc: swr-mstr-ctrl: Move to CPU-1 FIFO for swr register read/write
ipc: apr_vm: change the priority of apr_vm_cb_threa
asoc: get and put for TERT_TDM_TX_0 ADM channels
dsp: Reslove audiodlkm code warning
asoc: Reslove audiodlkm error & warn & issues
asoc: add drift support for mi2s interface
ASoC: add LXC audio config for auto sa8295 platform
asoc: enable mclk1 for audio bring up
ASoC: wsa883x: Enable all WSA interrupts
asoc: wsa883x: add slave_irq for wsa883x
asoc: wsa883x: update irq_drv_data per wsa device
ASoC: mbhc-adc: Prevent lineout from being reported two times
ASoC: Lahaina: correction to CPS payload
ASoC: wcd937x: Add change for LDOL VOUT accuracy calibration
ASoC: wcd937x: Update register value for new fab id for tanggu
ASoC: wcd937x: Add change to micbias reg for new tanggu fab id
Revert "ASoC: wsa883x: Add check before enable/disable PDM WD irq"
ASoC: Add proper routing for vi ch and cdc dma format widgets
asoc: set min period size of capture/playback to 120
dsp: bugfix for memory leak in error handling path
config: fix compile issue
asoc: Update the dai link IDs for MI2S dai links
ASoC: msm-dai-q6-v2: Add separate cpu dais for mi2s rx and tx
Change-Id: I0d9cf31f1ef60c754451ef86589cf91d29f8d682
* These symlinks were meant for the out-of-tree module. Now that we
build them in-tree, we need to correct them.
Change-Id: If2f506138c7e08123a90e0f897f0e899bb47ef0c
