Commit Graph

25516 Commits

Author SHA1 Message Date
e893772559 Add new owners to the test/ directory.
Add Artem Titov and Niels Möller as additional owners of test/ directory.

Bug: webrtc:10138
Change-Id: If195f7dfa892c34c3f727523777f1cd99b796fcb
Reviewed-on: https://webrtc-review.googlesource.com/c/117223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26234}
2019-01-14 10:55:37 +00:00
083fc3f7ed Adds nisse@ and sprang@ to test/OWNERS
Bug: None
Change-Id: If535cb41c128ccbb9e9550a2311645fadd44a2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/117222
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26233}
2019-01-14 10:20:03 +00:00
b2e21b014c Remove rtc_enable_android_opensl.
This GN argument is unused.

Bug: webrtc:10198
Change-Id: I470e3725758fc7d6e80673842fd36fa2f22339a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116993
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26232}
2019-01-14 10:00:40 +00:00
2bb29f018a Set callback_ member at start of desktop capturer Start()
Some callback wrappers set the callback_ member at the start, but
most set it after calling any owned implementation of Start().

Setting it after the call means that the callback_ is not set up
for any callbacks that happen during the call.

This cl fixes that by setting the callback_ member before any
calls are made in Start().

Bug: chromium:916961
Change-Id: Id26f8cc98377ef217f928095834162f5526c1fdf
Reviewed-on: https://webrtc-review.googlesource.com/c/117040
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
Cr-Commit-Position: refs/heads/master@{#26231}
2019-01-11 21:16:22 +00:00
b46235c1cc desktopCapture: skip non-responsive windows in the picker
This is a following up cl to the fix of crbug.com/911110. On Windows,
if an App window is suspended, it will block some queries (which
causes Chromium freezing and is fixed in Chromium.) and won't be captured.
So there is no reason to list it in the window capture picker.

Notes: this cl can't fix the case that the select app window becomes
non-responsive just before capturing starts. Hope that an extreme corner
case that can be safely ingored.

Bug: chromium:911110
Change-Id: I0d14872ac699d559f40b3bff70f048efc67ca5d9
Reviewed-on: https://webrtc-review.googlesource.com/c/115441
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26230}
2019-01-11 19:21:22 +00:00
977c82020c Rename AttachCurrentThreadIfNeeded to avoid clash with function.
A function with the same name exists here [1]. If the two headers are included
together this causes compilation errors.

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/sdk/android/src/jni/jvm.h?l=27&rcl=82f96e6a56e6230e98ee70de5178d7de69795c26

Bug: None
Change-Id: Icbc680f24a02ec66ea2b5e2b6584a53042cf45c7
Reviewed-on: https://webrtc-review.googlesource.com/c/116662
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26229}
2019-01-11 19:09:23 +00:00
07dc1e8594 (6) Rename files to snake_case: remove scripts and temp files
Tbr: kwiberg@webrtc.org
Bug: webrtc:10159
Change-Id: I8e3c8b0d42bffd85e8b582adb492523c9fb18eaa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/117026
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26228}
2019-01-11 17:17:04 +00:00
aec15aa810 (5) Rename files to snake_case: install forwarding headers
Mechanically generated with this command:

tools_webrtc/do-renames.sh install all-renames.txt && git cl format

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +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
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
1bab196551 (2) Rename files to snake_case: files to rename
rename-headers.txt: List of header files to rename.
    Generated first by find_header_renames.sh then
    curated by hand.

all-renames: List of all files to rename. Generated
    first by find_source_test_renames.sh with
    rename-headers.txt then curated by hand.

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ib6a56e440f62d9fb71964421c6533a66b3d3f1d2
Reviewed-on: https://webrtc-review.googlesource.com/c/115435
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26224}
2019-01-11 17:03:36 +00:00
5e130f05a0 (1) Rename files to snake_case: scripts
do-renames.sh: Take a list of files to rename and do
    perform the renaming (includes updating BUILD.gn,
    include guards, DEPS, include paths, and installing
    forwarding headers).

find_header_renames.sh: Looks through all header files
    and tries to guess what they should be renamed to,
    if they don't already have underscores.

find_source_test_renames.sh: Takes a list of header file
    renames and applies that information to renaming
    the corresponding source/test files.

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I073608e20bb163f3923ab2209eea72a115a4f593
Reviewed-on: https://webrtc-review.googlesource.com/c/91900
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26223}
2019-01-11 17:01:45 +00:00
aaa99a93e2 Add unittest for congestion window pushback in goog_cc.
Bug: none
Change-Id: Idc4ed71d8e12335eeaccbf1181eff36657f122d0
Reviewed-on: https://webrtc-review.googlesource.com/c/116320
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26222}
2019-01-11 15:42:57 +00:00
1f7a008261 Enable quality-scaling in all video perf tests.
Bug: None
Change-Id: Idc8d4b3372dcabdc4b419f1cce3d02adc3c30128
Reviewed-on: https://webrtc-review.googlesource.com/c/116983
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26221}
2019-01-11 15:39:17 +00:00
83953e4d95 Delete method StreamInterface::ReserveSize
Bug: webrtc:6424
Change-Id: I33d62599423b6c88c8e7117c347b7e0133d39943
Reviewed-on: https://webrtc-review.googlesource.com/c/116963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26220}
2019-01-11 14:08:45 +00:00
4687915495 Enable use of MediaTransportInterface for video streams.
Bug: webrtc:9719
Change-Id: I8c6027b4b15ed641e42fd210b3ea87d121508a69
Reviewed-on: https://webrtc-review.googlesource.com/c/111751
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26219}
2019-01-11 14:06:15 +00:00
7289906437 Delete enum NetEqDecoder.
A trimmed down version is moved to legacy_encoded_audio_frame_unittest.cc
where it's used for test parameterization.

Bug: webrtc:10185
Change-Id: I9abda22f9806b831b6ca4b27d6bcc888285f50f2
Reviewed-on: https://webrtc-review.googlesource.com/c/116961
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26218}
2019-01-11 13:52:25 +00:00
6c4b1b7ade Avoid depending on testonly target in event_log_visualizer_utils.
This is done by creating a custom ReplacementAudioDecoderFactory.

Bug: webrtc:8396, webrtc:10080
Change-Id: Ie1cb614fec855b82d65c6ef86c3593f547254559
Reviewed-on: https://webrtc-review.googlesource.com/c/116795
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26217}
2019-01-11 12:55:50 +00:00
ba50223363 Make voiceengine/audio transport stop using voice_detection() interface
Configuration for AudioProcessing::voice_detection() is moving into
AudioProcessing::Config, to get rid of the pointer-to-submodule
interfaces (such as voice_detection()).

Bug: webrtc:9947
Change-Id: Ia64ae996a43d44423aa0d612a3f1185b52a3e534
Reviewed-on: https://webrtc-review.googlesource.com/c/116067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26216}
2019-01-11 12:31:29 +00:00
b8b3c9918f Clean up visibility and dependencies of RTC event log build targets.
- Remove visibility of encoder target.
- Remove unnecessary dependency on task_queue.
- Remove CreateRtcEventLogFactory() declaration from the rtc_event_log_api target
  since the function is not defined in that target.

Bug: None
Change-Id: Id9edee86f358d08ea063d62bd96e9653c5b06d55
Reviewed-on: https://webrtc-review.googlesource.com/c/116060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26215}
2019-01-11 11:05:12 +00:00
e4ed6ea63b Introduce stats calculator.
It accumulate sample values inside it and provide API to calc
min/max/avg and percentiles. Current implementation will do it
in O(nlogn) time and planned to be used in the test code after
all time sensitive operations and also assume not too big amount
of data inside.

Bug: webrtc:10138
Change-Id: I262c4b9ca538c19463888b6d6bcdaa7e8c3caa68
Reviewed-on: https://webrtc-review.googlesource.com/c/116284
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26214}
2019-01-11 10:53:53 +00:00
6bb5ab9740 Reland "Refactor and remove media_optimization::MediaOptimization."
This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499.

Reason for revert: This change seemed innocent after all, so undoing speculative revert.

Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
> 
> This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
> 
> Reason for revert: Speculative revert due to downstream crashes.
> 
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> > 
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> > 
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> > 
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> > 
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> > 
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
> 
> TBR=nisse@webrtc.org,sprang@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
2019-01-11 10:46:33 +00:00
9c843906ca Delete VCMEncodedFrame methods Buffer and MutableBuffer
Replaced by inherited method EncodedImage::data().

Bug: webrtc:9378
Change-Id: I4ec75148f578c72ffb407f9cbf6b4232cc9cfcf6
Reviewed-on: https://webrtc-review.googlesource.com/c/116962
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26212}
2019-01-11 10:10:12 +00:00
87da937789 Delete unused constant kVideoCodecI420
Followup to cl https://webrtc-review.googlesource.com/c/112596.

Bug: webrtc:5791
Change-Id: Ie0375fa9e47dddd9e78d26fd63b8a349bacf5903
Reviewed-on: https://webrtc-review.googlesource.com/c/114983
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26211}
2019-01-11 09:04:56 +00:00
1daa7e8729 Use RTP timestamp when checking for frame duplication.
Value of render timestamp can be the same for consecutive frames (e.g.
when old frames got decoded and need to be rendered immediately). It
should not be used for frame duplication checking.

Bug: b/122636276
Change-Id: Ie00bdd3fa50a2eacd48cba228fa3c54a6b206864
Reviewed-on: https://webrtc-review.googlesource.com/c/116790
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26210}
2019-01-11 08:52:13 +00:00
6670a9d145 AEC3: More efficient comfort noise generation
Comfort noise was generated by picking random angles on the unit circle
for each frequency band and then obtaining points on the unit circle from
{cos(a), -sin(a)}.

In order to reduce complexity, this change introduces a randomly indexed
table of 32 elements over sin(a). cos(a) is obtained by adding an offset
corresponding to pi/2 to the index. The table is pre-scaled by sqrt(2) to
avoid later multiplications.

This change reduces the computational complexity of AEC3 by ~8% with no
audible degradation.

Bug: webrtc:10189
Change-Id: I8cfe2469022fb1fe910ab3f966e55d9d499b7161
Reviewed-on: https://webrtc-review.googlesource.com/c/116787
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26209}
2019-01-11 08:46:05 +00:00
0554368eed Delete method DecoderDatabase::RegisterPayload(...NetEqDecoder...)
Bug: webrtc:10185
Change-Id: I69ce40b1c7267b039cd1d2237c5d5bbae3a81875
Reviewed-on: https://webrtc-review.googlesource.com/c/116683
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26208}
2019-01-11 07:39:45 +00:00
0486f0914f Roll chromium_revision 8c0cc38022..783044b798 (621736:621838)
Change log: 8c0cc38022..783044b798
Full diff: 8c0cc38022..783044b798

Changed dependencies
* src/base: 155eaadd00..992951c2bf
* src/build: 67630827e1..19c19422cd
* src/ios: a2ac2bd4c2..40f164ac1e
* src/testing: d1c310b6d6..e2343647af
* src/third_party: 2da31fafff..9132ba856f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/775ce3b01f..0cc582388f
* src/tools: 7ebeeeb997..f1f7eab58d
DEPS diff: 8c0cc38022..783044b798/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I40b7ca062e1810cacd6d88bf715397477b193454
Reviewed-on: https://webrtc-review.googlesource.com/c/116900
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#26207}
2019-01-11 01:38:22 +00:00
395d29944a Roll chromium_revision 2e2ded718a..8c0cc38022 (621632:621736)
Change log: 2e2ded718a..8c0cc38022
Full diff: 2e2ded718a..8c0cc38022

Changed dependencies
* src/base: c4e5b7ca9d..155eaadd00
* src/build: 7b20546cf8..67630827e1
* src/ios: cd569bf30b..a2ac2bd4c2
* src/testing: e091f08842..d1c310b6d6
* src/third_party: 695a8d6bb4..2da31fafff
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/daced929e5..775ce3b01f
* src/third_party/depot_tools: b1be3782a4..80a1cf66b8
* src/tools: 043d1c8fe4..7ebeeeb997
DEPS diff: 2e2ded718a..8c0cc38022/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I1e01ebad4cd94de0b04c73a97d09dec2b2bab89b
Reviewed-on: https://webrtc-review.googlesource.com/c/116860
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#26206}
2019-01-10 21:23:53 +00:00
8578daf4b2 Roll chromium_revision 6e83997c8b..2e2ded718a (621525:621632)
Change log: 6e83997c8b..2e2ded718a
Full diff: 6e83997c8b..2e2ded718a

Changed dependencies
* src/base: a6d274ed72..c4e5b7ca9d
* src/ios: 40796f6970..cd569bf30b
* src/testing: cb05f60e96..e091f08842
* src/third_party: 74fc63bb69..695a8d6bb4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/79517a0b03..daced929e5
* src/tools: cb83ff7a00..043d1c8fe4
DEPS diff: 6e83997c8b..2e2ded718a/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I2a81ac312c6d73c7268be2de21dcba9fc0557f8d
Reviewed-on: https://webrtc-review.googlesource.com/c/116821
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#26205}
2019-01-10 18:44:03 +00:00
e9bece37ae Minor change to the Json Config format for the replay file.
See: test/fuzzers/configs/replay_packet_fuzzer for example configurations.

Bug: webrtc:10117
Change-Id: Ife2bf7d053bc4feb4d7e6e38ff31280236c962b6
Reviewed-on: https://webrtc-review.googlesource.com/c/116764
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26204}
2019-01-10 18:38:08 +00:00
53eae87bf8 Add PeerConnection option to enable RTX handling in the audio jitter buffer.
Bug: webrtc:10178
Change-Id: I70abce0c7b74124d2b1978d9a5eb8216b6233d1a
Reviewed-on: https://webrtc-review.googlesource.com/c/116784
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26203}
2019-01-10 16:28:43 +00:00
43f3982d6f Remove TaskQueue::PostAndReply as unused
Bug: webrtc:10191, webrtc:9728
Change-Id: Iaaa7c88bbbbfdd6e3e9bf5ab683bbdb2962a5cab
Reviewed-on: https://webrtc-review.googlesource.com/c/107202
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26202}
2019-01-10 16:06:57 +00:00
a8f58f001e Add data() accessors to EncodedImage
Intend to make the |_buffer| member private, in a later cl.

Bug: webrtc:9378
Change-Id: I8398932a36d8d931a7e587edca7be3957bbafcfd
Reviewed-on: https://webrtc-review.googlesource.com/c/116782
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26201}
2019-01-10 15:30:55 +00:00
6551faf089 Refactor FrameBuffer to store decoded frames history separately
This will allow to increase the stored decoded frames history size and
optimize it to reduce memory consumption.

Bug: webrtc:9710
Change-Id: I82be0eb376c5d0b61ad5d754e6a37d606b4df29d
Reviewed-on: https://webrtc-review.googlesource.com/c/116686
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26200}
2019-01-10 15:11:15 +00:00
0a7d56e0e5 Delete method StreamInterface::GetSize
Followup to https://webrtc-review.googlesource.com/c/4821

Bug: webrtc:6424, webrtc:7811
Change-Id: I6a4d8b52937256832509ebd33123c7b004263d8f
Reviewed-on: https://webrtc-review.googlesource.com/c/101181
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26199}
2019-01-10 15:04:04 +00:00
b344640771 Enable quality scaling in video_loopback.
Bug: None
Change-Id: Ie6e7472f8b407b7da0f111cddec35bbbe66e31df
Reviewed-on: https://webrtc-review.googlesource.com/c/116791
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26198}
2019-01-10 15:02:03 +00:00
1bdce799eb Parse logs without RTX SSRC even if there is an RTX payload type.
Bug: webrtc:10187
Change-Id: I8f446ce5a8960fdaa6e3193c6647b8133b63e9a7
Reviewed-on: https://webrtc-review.googlesource.com/c/116741
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26197}
2019-01-10 14:43:39 +00:00
69321ddfb5 Make FrameBuffer support an unlimited number of dependents per frame
Bug: webrtc:10190
Change-Id: I59680ec0dc05bc77dcbef50ddbb83ce2bcd91f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/116788
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26196}
2019-01-10 14:36:47 +00:00
977cd17316 Make VCMDecodeErrorMode optional when calling VideoCodingModule::SetReceiverRobustnessMode
This is a preparation for deleting other modes than
VCMDecodeErrorMode::kNoErrors.

Bug: webrtc:8064
Change-Id: I614f8012f306c5d59e72bdb851b582c286cdd130
Reviewed-on: https://webrtc-review.googlesource.com/c/116781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26195}
2019-01-10 14:06:10 +00:00
2874672671 Delete method VideoCodingModule::SetVideoProtection
Bug: webrtc:8064
Change-Id: I2a6ed11bf1415e4e0d199733f9d9a659afec0fe8
Reviewed-on: https://webrtc-review.googlesource.com/c/116689
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26194}
2019-01-10 13:32:07 +00:00
f7d636644f Delete method NetEqImpl::CurrentDelayMs, used only by tests
Bug: None
Change-Id: If94695f60ed804f6b43be828dd93f02826269140
Reviewed-on: https://webrtc-review.googlesource.com/c/116687
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26193}
2019-01-10 12:49:12 +00:00
8984cd61ca Revert "Add a high bitrate full stack test with fake codec"
This reverts commit 15df2774f4e85cf8900768c1793edcf17d651dcd.

Reason for revert: It's causing the Android perf bots to fail. E.g.: https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20%28L%20Nexus4%29/6666

Original change's description:
> Add a high bitrate full stack test with fake codec
> 
> This CL adds a fake codec factory  in WebRTC that can be used in tests to
> produce target bitrate output.
> 
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
> 
> Bug: chromium:879723
> Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
> Reviewed-on: https://webrtc-review.googlesource.com/c/97185
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26182}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org

Change-Id: I33cd01ce345d81d66543f9be99750fa100760b5c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:879723
Reviewed-on: https://webrtc-review.googlesource.com/c/116785
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26192}
2019-01-10 11:49:05 +00:00
6613f8e98a Revert "Refactor and remove media_optimization::MediaOptimization."
This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.

Reason for revert: Speculative revert due to downstream crashes.

Original change's description:
> Refactor and remove media_optimization::MediaOptimization.
> 
> This CL removes MediaOptmization and folds some of its functionality
> into VideoStreamEncoder.
> 
> The FPS tracking is now handled by a RateStatistics instance. Frame
> dropping is still handled by FrameDropper. Both of these now live
> directly in VideoStreamEncoder.
> There is no intended change in behavior from this CL, but due to a new
> way of measuring frame rate, some minor perf changes can be expected.
> 
> A small change in behavior is that OnBitrateUpdated is now called
> directly rather than on the next frame. Since both encoding frame and
> setting rate allocations happen on the encoder worker thread, there's
> really no reason to cache bitrates and wait until the next frame.
> An edge case though is that if a new bitrate is set before the first
> frame, we must remember that bitrate and then apply it after the video
> bitrate allocator has been first created.
> 
> In addition to existing unit tests, manual tests have been used to
> confirm that frame dropping works as expected with misbehaving encoders.
> 
> Bug: webrtc:10164
> Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26147}

TBR=nisse@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10164
Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
Reviewed-on: https://webrtc-review.googlesource.com/c/116780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26191}
2019-01-10 11:39:24 +00:00
e449805f42 APM unit test: echo path gain change events notified.
This CL adds two unit tests to make sure that, when an echo path gain
change occurs, the echo canceller is notified.
Such a change can be caused by (i) a pre-amplifier gain change or
(ii) an analog gain change.

Bug: webrtc:7494
Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d
Reviewed-on: https://webrtc-review.googlesource.com/c/111780
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26190}
2019-01-10 11:06:24 +00:00
949099d50d Roll chromium_revision 4a7f224d18..6e83997c8b (621417:621525)
Change log: 4a7f224d18..6e83997c8b
Full diff: 4a7f224d18..6e83997c8b

Changed dependencies
* src/base: 39a3ef888a..a6d274ed72
* src/ios: 26e8a7f01b..40796f6970
* src/testing: 98928f83e8..cb05f60e96
* src/third_party: 1f39c2b117..74fc63bb69
* src/third_party/depot_tools: 2d4a955e90..b1be3782a4
* src/tools: d14e904200..cb83ff7a00
DEPS diff: 4a7f224d18..6e83997c8b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I968e22c9d82311e2d4257c056e82d7d3b2ef9558
Reviewed-on: https://webrtc-review.googlesource.com/c/116771
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#26189}
2019-01-10 10:52:19 +00:00
5dcadff258 Fix macOS demo privacy crash.
The macOS demo's Info.plist doesn't contains camera and microphone usage description, which will cause demo crash when starting call.

Bug: none
Change-Id: Ie85b0087e6aa6e768a8e6740fffe0b95891b20dd
Reviewed-on: https://webrtc-review.googlesource.com/c/116703
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26188}
2019-01-10 10:50:49 +00:00
39b934ba2e Add NetEq config flag that enables RTX handling.
When enabled, the delay manager is updated with reordered packets. It also makes the peak detector ignore the reordered packets.

Change-Id: I2bdc99764cc76b15e613ed3dc75f83aaf66eee4e
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/116481
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26187}
2019-01-10 10:04:34 +00:00
d8bd75079b Add ability to use movable only functors in rtc::Thread::Invoke(...)
Add support for movable only functors with void return type. Non void
return type is already supported.

Bug: webrtc:10138
Change-Id: If2ae2b5ab7244a0e932bceff7d9853c030805688
Reviewed-on: https://webrtc-review.googlesource.com/c/116740
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26186}
2019-01-10 09:11:48 +00:00
862025352b Minor changes to TestVideoReceiver.
Moved common code to helper method.

Bug: none
Change-Id: Iafae1a6e96c9d38cab8dd7d410d9f8717ee1ecb2
Reviewed-on: https://webrtc-review.googlesource.com/c/91862
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26185}
2019-01-10 08:15:28 +00:00