| Commit message (Collapse) | Author | Age | Files | Lines |
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-Fix compilation errors from qahw_api tests and enable
compilation of qahw_api on Android builds.
Change-Id: Ifc05ea22f0fd9ac0e6f6e9b22a6c593841acc761
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Add liblog to the dependency list for voice_processing and
disable compilation of audiod when AOSP flag is set.
Change-Id: Ie60559806a3ee02e6b84626ba2a3ee89844a1a7a
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- Initialization of 'realtime' mode for an input stream got
moved down its usage during propagation of compress input
changes to 2.2 component. Incorrect mode causes legacy low-latency
configuration being used even for NOIRQ pcm devices and hence
the error in setting hw and sw parameters.
- Realign initialization of 'realtime' mode to place it properly
before the usage and hence fix this issue.
Change-Id: Ifdc7b8ff9d129e93a48d34c622e5a76bd83617eb
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audio-userspace.lnx.2.2-dev
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-Asrc mode is not getting set on headphone device for ringtone
playback with concurrent native DSD or 44.1 Khz playback.
Check and set ASRC mode is done in select_devices function where
it has combo device for ringtone playback and hence it does not set
asrc mode.
-Invoke check_and_set_asrc_mode API from enable_snd_device where
combo device is split.
CRs-Fixed: 1106679
Change-Id: I2e79dcffa8022532801c0feb7d527de81a885001
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into audio-userspace.lnx.2.2-dev
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-Speaker backend also gets configured to HIFI configuration, if
a combo request comes in during any HIFI playback.
-This leads to failure in playback on speaker.
-Add a separate check to ensure that for any speaker backend
the configuration is fixed to correct values.
Change-Id: I9f383438c2f10804b093505ff605eee7221a7fb5
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Add support for AAC LATM format in hal layer
Change-Id: I1314fb8759a59845cd7cba879f829d6fe2a0f53c
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- In case of speaker protection, capture usecase is not having
valid input stream associated with it. Hence attempt to access
stream info while sending calibration may result in segfaults.
- Add check in send calibration, to use stream info only if its
available, and fallback to defaults if stream is not set.
Change-Id: I880f8c66e6cba65947846dd2fef74a74be125aa9
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Make sure if the TARGET_USES_AOSP flag is defined
then all QTI specific features should be disabled
and compilation is successfull.
Change-Id: I440b538b5449177e14ca47f1dcfbec70a2ee7fed
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active" into audio-userspace.lnx.2.2-dev
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Device configuration mismatch is observed when VOICE and VOIP are active.
When VOICE and VOIP usecases are present, and device switch to speaker,
VoIP snd_device is set to speaker, but the related ACDB configuration
is not updated correctly.
Fix this by update voc cal data for VoIP when new device is selected.
CRs-Fixed: 1086922
Change-Id: I6655773af12a63acc302d2c9eeeace289cb46f70
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Enable Compress passthrough
Add Dolby and DTS formats which got removed mistakenly before.
CRs-Fixed: 1110685
Change-Id: I20f42b14bafb9b1a36bf471037f4e462e8cbd3cb
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- Add sdm660 qrd skus variant sound card
details
CRs-Fixed: 1088368
Change-Id: Ife20f7975d936bd32a9d484a8c58884320dba9d7
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audio-userspace.lnx.2.2-dev
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- Adding control for compress output format
- Fix for Avsync issue whild playing video clip due to QAF latency.
- updating to use channels instead of channel_mask.
- Added support for msmd
- Added Passthrough playback support for AC3, EAC3 and
Multichannel PCM formats.
Change-Id: Ia52192b6a8a9970617d0f1d9b3f613d12beb73c6
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-Validate active_input before accessing it, it NULL use
default value.
-For voice call scenario it is not necessary to have an active
input stream.
Change-Id: I258b9e8c098d76f838bcca5b968d64369243885e
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When voice call happen during compress VOIP call, VoIP input would
choose audio record when insert headset. First when voice call is
started, the call mode change to IN_CALL. Second compress VOIP input
stream would be closed, and then re-open when headset insert. Due to
call mode is IN_CALL, the input stream only can use audio-record
usecase even if compress VOIP still is active.
This change is to make sure VOIP input and output use VOIP path
when compress VOIP call is active.
CRs-Fixed: 1087113
Change-Id: If0c8151197b3617b112383d97dea5a290525bb0b
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Close mixer opened by platform during deinit to avoid fd leak.
This mixer fd can be accumulated in a prcocess if hal is loaded-unloaded
several times.
CRs-Fixed: 1095695
Change-Id: Ib5a6f0628f178f897cd67d80f32de7e3515aa9f0
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audio-userspace.lnx.2.2-dev
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update active_input only after the usecase that is being
stopped is removed from the usecase list. Otherwise,
active_input can be set to the same usecase that is being
terminated and input stream deallocated. At that point
members of active_input will be invalid.
CRs-Fixed: 1094825
Change-Id: I200ed352ff3c9d048884c9e464ba7d75fc7beeca
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Add new set api to set Sourcetracking parameters.
Change-Id: Ib166c521a2f48b940798a760fd323eb076caa143
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audio-userspace.lnx.2.2-dev
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Initial version of QTI audio effect HAL Wrapper.
CRs-Fixed: 1081403
Change-Id: I12291cc7106f7530422891d1bee7e3d4530563c5
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Playback over USB headsets fails as the default backend index
is selected.
Return the correct backend index by adding a check for USB headset
to fix the issue.
Also backport a few bug fixes from USB tunnel mode testing on 8953.
CRs-Fixed: 1105780
Change-Id: If923f486a44070a4824b80199bcaed5d902462fe
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Input device for Voice usecase do not update proper when VOIP
is closed. When IN_COMM mode is set, and VoIP do not start up,
input device selection for Voice call cannot get proper snd_device.
Fix this by check in_call flag to ensure input device update proper
CRs-Fixed: 1101637
Change-Id: I30d594e0c15c0265958a313ce09748d156c7c750
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When SSR is happening and the sound card is offline, after removing
an offload usecase from the usecase list, start traversing from the
beginning of the list on the next iteration. This must be done
because in the process of removing the current node from the list,
there is a moment in the time where another thread can remove the
node immediately after it which is also pointed to by tempnode
that is used by list_for_each_safe. This would result in tempnode
pointing to invalid memory on the next iteration of list_for_each_safe,
causing a segfault.
CRs-Fixed: 1088561
Change-Id: I8a220afa341c4e62612a809064754ddcd497d88a
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Add support for AFE sidetone for USB and non-USB
devices when device sidetone is not available or
not supported.
Change-Id: Iaacaa61fc1eddacb1fe9d58ef194d8f980d8f934
Signed-off-by: Vikram Panduranga <vpandura@codeaurora.org>
CRs-Fixed: 1061420
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-To ensure that A2DP stream is suspended before SCO starts, force
route to speaker is done, update the logic to ensure that force
routing to speaker only happens when there is no A2DP device active.
-Ensure that a2dp_start retry happens on routing request if the earlier
start failed.
Change-Id: I29ea0d8857fc4b9d837f9954423861be9b43b9a6
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Assign the backend mixer control based on interface.
Add support to set channels for afe backend for playback.
Change-Id: Ia870cb51b9e76700ef67812ee9af1437c76bf18c
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Update path of mixer_xml_wcd9335 for LE variant.
CRs-Fixed: 1105335
Change-Id: I3265ac9f2f17a821d2f8476d4050ed84f9b1170c
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Enable afe proxy feature flag for a2dp src to work
CRs-Fixed: 1081411
Change-Id: I30672f5549c49fcd51ffd6ceed2a11c366e7562f
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Update the code name from msmfalcon to sdm660. As part of this, update
the filename containing "falcon" and files content containing "falcon".
Conflicts:
configs/sdm660/sdm660.mk
Change-Id: I5fc11ffac2f21f11e2a7283cf7375bcf93c01623
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audio-userspace.lnx.2.2-dev
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-Fix compilation warnings against tx app_type changes.
-Remove assignment of compress usecase outside compress
input extension.
-Define input flag for compress record usage.
Change-Id: I36630c7f6524d494b76e0a3bd074185a01e19883
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audio-userspace.lnx.2.2-dev
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Add new input profile to support 24 bit record.
Based on bitwidth select appropriate unprocessed mic device.
Change-Id: Iba7cdf733f6993129cea1bf41908831ac1dda8d4
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Relay the device connect/disconnect events to sound trigger hal
in order to support voice activation using mic devices other
than handset mic.
Change-Id: Id4b186269f6f39ebbba605f4819965365b33b7e7
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- Add support for MP2, Dolby and DTS formats
- Correct the value of compress passthrough flag
Change-Id: Ia909f17eb3354893fac41dc31ac3b50bf2e3ef08
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Add MP2 as a valid audio format.
Fetch correct SND CODEC ID for MP2 format.
Change-Id: I187b305a522581b6510160541e7b15ae13f08d66
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Update the mixer paths for LE build due to the paths different from
LA build.
CRs-Fixed: 1098884
Change-Id: Ifa3e049ad70cbab9386c9dec99cd9d8074a9edf5
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Update Makefile and Configure file to compile
target specific platform file.
Change-Id: Ic2e5c3f5cc5484d2acaacb9708595ee670d53c03
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Add sample rate as a part of argument in gef callback, as client needs
this information to determine what's the exact sample rate COPP is
running and get calibration value accordingly.
If there's gef APIs call in callback, device lock will be probed twice.
Temproraily unlock device lock around GEF device change notification
to solve deadlock.
CRs-Fixed: 1094022
Change-Id: I47fdb1f88397e4340ba930113c64c140596bc62a
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Do not allow configurable mono speaker assignment for devices with
one speaker. For msm8998, avoid it for fluid variants.Also, fix typo
in selection of VOICE_SPEAKER_2_PROTECTED device during send_voice_cal.
CRs-Fixed: 1082853
Change-Id: I7124c7ca7424feea1f2e54bcb6bc581313df6259
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- FM playback is broken since incorrect pcm id is used.
- Fix by adding the missing usecase to the usecase index table.
Change-Id: I63b00702ddadb816c4c5d42548b47b1426bddc20
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- Add msmfalcon target variant sound card
details.
Change-Id: I9b5b281a9ae4209fe163325d889ff248af13e91f
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Limit length of the property name to adhere to property max length
CRs-Fixed: 1770914
Change-Id: Iba244dd21fc6ba00fd0f7935c59ff258795316b8
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