Age | Commit message (Collapse) | Author |
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Add support for 352800 sampling rates and 32 bps for the TDM.
Add support for PCM compress passthough mode with mixer controls.
CRs-fixed: 1116515
Change-Id: Iab059a5a6b6ce8f57717023467677a399a60032e
Signed-off-by: Josh Kirsch <jkirsch@codeaurora.org>
Signed-off-by: Karthikeyan Mani <kmani@codeaurora.org>
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Afe driver changes to query avtimer vs device drift.
Drift obtained can be used to pull the device pll so
that avtimer and device are in sync.
CRs-Fixed: 1112258
Change-Id: I4d4ddb0dbc06270553d583f266a44ddbe9412d1a
Signed-off-by: Manish Dewangan <manish@codeaurora.org>
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Add mixer control to set the endianness of the playback/capture
USB device.
CRs-Fixed: 2003737
Change-Id: I99102c3bb64e321fb3e5df38428e63a406f91d7e
Signed-off-by: Haynes Mathew George <hgeorge@codeaurora.org>
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Add support for multi-copps for multiple BEs with independent
calibration data. This allows for more accurate calibration of device
copps.
CRs-fixed: 1110411
Change-Id: I72ce501408a474eb620a088172e3c4d789ab5ef0
Signed-off-by: Siena Richard <sienar@codeaurora.org>
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Enhance LSM code to support maximum configuration of 48khz, 24bit
and 4 channel.
Change-Id: I03895c983527d87389ca69e85235b1def5b4a2fa
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Revathi Uddaraju <revathiu@codeaurora.org>
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Add support to parse LSM_SESSION_EVENT_DETECTION_STATUS_V3
in cases where event status requires timestamp corresponding
to detection.
Framework mode config is set to timestamp mode in cases where
LSM_SESSION_EVENT_DETECTION_STATUS_V3 is required.
Change-Id: Id0da3b24d55ac56ff6b61372ede9c63f50b2f4d4
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Revathi Uddaraju <revathiu@codeaurora.org>
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Microphone2 and SND_JACK_BTN_5 enum are same, resulting
in SND_JACK_BTN_5 event getting triggered when ANC headset
is plugged into the device. Use unique values in
sound jack type enum.
Change-Id: I668e50afcad11b1f62f511f4241f79bad858c7d2
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
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LSM will connect to ADM to apply preprocessing and improve
detection performance. LSM can also directly connect to AFE
similar to the existing mechanism.
MAD polling will be disabled in case of LSM connects to ADM.
Add EC reference end channel, bit format and sample rate control
to configure far end params for Echo Cancellation.
Change-Id: I4684ae346884d656e95350b7a63929b91a843512
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Revathi Uddaraju <revathiu@codeaurora.org>
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Polling needs to be disabled when LSM connects to ADM.
Add interface to enable or disable polling through
set_params. Add support to set port.
Change-Id: If027418a6d8a1ea48dcb6a0c146f68e7dd7a2664
Signed-off-by: Chaithanya Krishna Bacharaju <chaithan@codeaurora.org>
Signed-off-by: Revathi Uddaraju <revathiu@codeaurora.org>
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Add changes to support aptx decoder in offload mode.
Add support to set BT device address and add new mixer control
to set license key.
CRs-Fixed: 1106128
Change-Id: Idd4ec8ab829883ef4848be8b686e24101ccbed60
Signed-off-by: Dhanalakshmi Siddani <dsiddani@codeaurora.org>
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The timestamp API is called even when timestamp flag is not set.
Correct the read API calls based on timestamp flag.
CRs-fixed: 1107319
Change-Id: Ic40b166e9ddd42f20fecadcd4eafe187b3ff8785
Signed-off-by: Satish Babu Patakokila <sbpata@codeaurora.org>
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Allow MIC channel selection as the primary channel requested
by the application on Multichannel Echo Cancellation module.
CRs-fixed: 1102004
Change-Id: I94ba0f3171b4c7ebcc9936ca4f192f68de7be034
Signed-off-by: Rahul Sharma <sharah@codeaurora.org>
Signed-off-by: Derek Chen <chenche@codeaurora.org>
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Add support for AFE sidetone whenever USB device or
codec sidetone is not available or supported.
Change-Id: I325ba6448efb4c021a2974ad813be4f8192e9ad1
Signed-off-by: Vikram Panduranga <vpandura@codeaurora.org>
Signed-off-by: Siena Richard <sienar@codeaurora.org>
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Add ASM low latency loopback support.
ICC (in-car communication) use case needs the low latency
ASM loopback to satisfy the latency requirement.
CRs-fixed: 998118
Change-Id: If225e809072f4296bc22028da29e589137e5799d
Signed-off-by: Derek Chen <chenche@codeaurora.org>
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Add support for AAC LATM clips. Define macro for AAC LATM format
and configure media format block with this value for AAC LATM clips.
CRs-Fixed: 1108268
Change-Id: I10cf2db9fc7e4d40d2bf910e162b3c2d95d8a03c
Signed-off-by: Surendar karka <sukark@codeaurora.org>
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Add control via ADM set PP Params to control volume of ICC
through a mixer command "Internal ICC Volume".
CRs-fixed: 1025376
Change-Id: I2b7099fe6d3a510859af42f1ac37a6db6e1b453c
Signed-off-by: Derek Chen <chenche@codeaurora.org>
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Add support for ADM_CMD_DEVICE_OPEN_V6 when the mixer path
specifies to use the multi-mic-echo-reference, which configures
the EndPoint2 for Echo Reference.
CRs-fixed: 1022080
Change-Id: I474f39a3437fa18003f4342e003d689b95837699
Signed-off-by: Derek Chen <chenche@codeaurora.org>
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Extend the media format options passed to the voice driver to
include sample rate, bits per sample, and channel mapping in
addition to port ID and the number of channels. Additional media type
information is provide to help avoid additional buffers required in
case of non-fractional sample rate for post processing voice data.
CRs-Fixed: 1065881
Change-Id: Ib69b57dc677b87fecfd689df7f8fc7ec8b4bc59f
Signed-off-by: Siena Richard <sienar@codeaurora.org>
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Add a new dailink for slimbus VI sense recording.
SLIMBUS_TX_VI and SLIMBUS_4_TX can coexist.
VI recording and speaker protection cannot work
simultaneoulsy due to shared physical afe port.
CRs-Fixed: 1087025
Change-Id: If074b7042e09d1e69147546461e6fa42d427350e
Signed-off-by: Xiaojun Sang <xsang@codeaurora.org>
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Add timestamp support for compress driver.
Modify ASM driver to read the buffer from predefined offset.
CRs-fixed: 1072067
Change-Id: I1c46befc223285495b3c0650e6c3eaae81f58771
Signed-off-by: Satish Babu Patakokila <sbpata@codeaurora.org>
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In common clock framework, API for registering clk
provider for a node is changed to of_clk_add_provider.
Add new audio ext clock file to use new API to register
audio clk provider.
Crs-Fixed: 1090500
Change-Id: I1d7ecc6a3d4c48d0de9645043d5e5bfdfa1d1f5f
Signed-off-by: Meng Wang <mwang@codeaurora.org>
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Add support in audio stream manager to configure decode
of g711 related clips.
CRs-Fixed: 1094107
Change-Id: Ie90fd68e24e7e793aaac64290e3c1e41682d6d5a
Signed-off-by: Yamit Mehta <ymehta@codeaurora.org>
Signed-off-by: Surendar karka <sukark@codeaurora.org>
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Add support in audio stream manager to configure encode
of g711 format
CRs-Fixed: 1094107
Change-Id: I496a975e427f68d7bb5cc2789bfc9bb949658233
Signed-off-by: Surendar karka <sukark@codeaurora.org>
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Add support for new DSP INTx MCLK, INTx IBIT CLK
and INTx MI2S ports. New internal codec and msm
based soundwire codec use these clocks and port_ids.
CRs-Fixed: 1083537
Change-Id: I72e0a15c8a283c68a3ed10cfd02a4e3d9526c312
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
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In combo usecase there are 2 front-end dai's with
same codec dai, for example, multi-phrase ADSP SVA detection.
Using a single bit as the counter causes the counter to roll
over to 0 during combo usecase.
To resolve this, change counter to unsigned int from single bit.
CRs-Fixed: 1086127
Change-Id: I2dd07bd967b7d4fb4878b6d65bd0f011c6b15bdd
Signed-off-by: Walter Yang <yandongy@codeaurora.org>
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It is possible that codec hardware can be reset in case of subsystem
restart scenarios. It is required to reset the codec DSP as well in
such cases to make sure the DSP is in usable state after the codec
hardware is reset. Change adds support to handle codec down and up
events and perform the necessary reset on codec DSP.
Change-Id: I79502c043f5e16947c895aab7cd584d72ad1a7dc
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
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The wdsp_intr_handler is not really processing interrupts in interrupt
context, rather assumes that the caller is not in interrupt context and
performs calls that may sleep. Rename the function in order to avoid
confusion. Interrupt handlers can still call this function as long as
they are in threaded interrupt context and are okay to sleep.
Change-Id: Ia2803d6ca021d505ed2e711e676cbd701b11c492
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
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command"
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Add support for ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4
command. This command adds support to playback/record 32 bit data
in 32 bit word and also provides a way to inform DSP about the
endianness of the data.
Change-Id: I3b013bedde8ccfa97a02e255e237df0cf2de13b8
Signed-off-by: Ashish Jain <ashishj@codeaurora.org>
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Add support for configurable bit width for AFE encoder.
Add new mixer ctl to set for usecases which enables
configuring different input/output bit format
on AFE for usecases such as APTXHD encoder for 24bit input
and 16bit output.
Change-Id: I62326a097cbd71a3ec2b93a0120284d8f71f5d57
Signed-off-by: Naresh Tanniru <ntanniru@codeaurora.org>
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It is possible that wcd codec DSP can crash or become unresponsive.
During such case, an error interrupt is generated by the codec. Add
support in manager driver to handle this interrupt and perform
subsystem restart to shutdown and reboot the DSP so that the DSP
can be recovered from the crash.
CRs-fixed: 1071949
Change-Id: I4662b5120bf7f731e399a27d8a613e2f3b648b00
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
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The codec driver is notified through interrupt whenever the
DSP in codec is unresponsive/crashed. In such cases, collecting
dumps of the codec memory is useful in debugging the issues.
Change adds support in the wcd dsp manager driver to collect
the memory dumps upon notification of error interrupt.
Change-Id: Ib91cd4fc1476ee1a9ec448cde1a083070443f726
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
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Currently, when manager driver interrupt handler callback is called,
there is no way for the caller to provide any data information along
with the type of interrupt. Change adds argument to the interrupt
handler so that callers can use this to pass agreed data when an
interrupt occurs.
Change-Id: I1c049227875a802491e21998c13c0bcd8eab7de6
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
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AAC encode is failing in ADSP due to mismatch
in channel config datatype.
Update channel config data type for aac encoder.
Change-Id: I844d6e1ac1b2b171cd74a2601ae09280a22589c9
Signed-off-by: Naresh Tanniru <ntanniru@codeaurora.org>
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