1c41be6e05
Propagate TaskQueueFactory to AudioDeviceBuffer
...
keep using GlobalTaskQueueFactory in android/ios bindings.
Switch to DefaultTaskQueueFactory in tests.
Bug: webrtc:10284
Change-Id: I034c70542be5eeb830be86527830d51204fb2855
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 08:00:49 +00:00
185e802971
Prefix AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO with WEBRTC_.
...
Since it is a WebRTC-only macro, let's prefix it with WEBRTC_.
Bug: None
Change-Id: I309666858ea898dc7cd1a68c21be190f98c87b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129935
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27327}
2019-03-28 08:44:27 +00:00
d970807e0c
Remove rtc_base/scoped_ref_ptr.h.
...
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o .
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
2fd09a40af
Remove deprecated code from audio device.
...
Bug: webrtc:7306, webrtc:10198
Change-Id: Iaeef4d7449c18325511f1763eba510b385959bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/118446
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26383}
2019-01-24 11:27:38 +00:00
10542f21c8
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
...
Mechanically generated by running this command:
tools_webrtc/do-renames.sh update all-renames.txt && git cl format
Then manually updating:
tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
140b1d94dc
Eliminate use of EventWrapper from android audio device tests
...
Bug: webrtc:3380
Change-Id: I746d2245966afe89065472d4a6a7447f8c63f9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/110163
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25598}
2018-11-12 13:22:46 +00:00
cfbd26df1e
Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC
...
First version was reverted in https://webrtc-review.googlesource.com/c/src/+/97941 .
The issue is now fixed.
TBR=ivoc
Bug: b/113648245
Change-Id: If631fdea95aa963952f15e48e9d2d678797dc225
Reviewed-on: https://webrtc-review.googlesource.com/97942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24573}
2018-09-05 10:24:35 +00:00
e2924d555d
Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."
...
This reverts commit f217903a67995496a1d67674d77d5f237772b01b.
Reason for revert: Breaks downstream tests
Original change's description:
> Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
>
> Also ensures that audio parameters are accessed atomically.
>
> Bug: b/113648245
> Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
> Reviewed-on: https://webrtc-review.googlesource.com/97331
> Reviewed-by: Minyue Li <minyue@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#24550}
TBR=henrika@webrtc.org ,ivoc@webrtc.org ,minyue@webrtc.org
Change-Id: I620406f25762cf76db0470b3b29b50bc146935c7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/113648245
Reviewed-on: https://webrtc-review.googlesource.com/97941
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Commit-Queue: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24569}
2018-09-05 08:52:51 +00:00
f217903a67
Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
...
Also ensures that audio parameters are accessed atomically.
Bug: b/113648245
Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
Reviewed-on: https://webrtc-review.googlesource.com/97331
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24550}
2018-09-04 11:22:53 +00:00
665174fdbb
Reformat the WebRTC code base
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Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
883d00f7d1
Add support of AAudio in native WebRTC on Android O and above
...
Bug: webrtc:8914
Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde
Reviewed-on: https://webrtc-review.googlesource.com/56180
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:20:27 +00:00
3def74b1b8
Disables AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex on Android
...
TBR=henrik.lundin
Bug: webrtc:7744
Change-Id: I0ebfd3016d9d6b815d5b1801e8481363da11af54
Reviewed-on: https://webrtc-review.googlesource.com/7200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20179}
2017-10-06 10:19:44 +00:00
92ea95e34a
Fixing WebRTC after moving from src/webrtc to src/
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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf
Moving src/webrtc into src/.
...
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
