Commit Graph

26628 Commits

Author SHA1 Message Date
dcfe484f2e Remove definition of macro WEBRTC_THREAD_RR (it's unused).
Bug: None
Change-Id: Id0ba90502ca3acb9e44665fd4cf788679f2c6652
Reviewed-on: https://webrtc-review.googlesource.com/c/117163
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26246}
2019-01-14 15:01:38 +00:00
45340ca824 Remove legacy video codec factories.
Removes the deprecated video codec factories and the related flag and
helper classes.

Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
2019-01-14 14:56:40 +00:00
959e9b6b57 Publish rtc::QueuedTask in api as webrtc::QueuedTask
Bug: webrtc:10191
Change-Id: I7dcba28615c2f3e44442be410dedde15f5fb1deb
Reviewed-on: https://webrtc-review.googlesource.com/c/113502
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26244}
2019-01-14 14:48:12 +00:00
645df9e3b5 Introduce Y4mFrameReader.
Bug: webrtc:10138
Change-Id: I213a4309a8a4b1a7afd296bf45566c9b3f9a215c
Reviewed-on: https://webrtc-review.googlesource.com/c/117301
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26243}
2019-01-14 14:03:08 +00:00
a8c7326524 RefCounter::DecRef: Remove obsolete TODO and update comment
Bug: webrtc:10198
Change-Id: Icbcb849bdc789d9e3bb7ea6a902475a8263980bf
Reviewed-on: https://webrtc-review.googlesource.com/c/117300
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26242}
2019-01-14 14:01:36 +00:00
ebd9770b05 Add extra documentation and minor fixes into video quality analyzer
Bug: webrtc:10138
Change-Id: Ia4ce55ca5ff92237c8d58811df8fd96cd650a5b0
Reviewed-on: https://webrtc-review.googlesource.com/c/116685
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26241}
2019-01-14 13:27:02 +00:00
13717842df Introduce DecodedFramesHistory class and use it in FrameBuffer
This is a space efficient way to store more records about decoded frames,
which is needed for long term references.

Bug: webrtc:9710
Change-Id: I051d59d34a966d48db011142466d9cd15304b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/116792
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26240}
2019-01-14 13:09:39 +00:00
d0f0f68953 Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 1.
Replaced with a combination of cricket::FakeFrameSource and
webrtc::test::FrameForwarder. This cl converts the first three
affected tests, the rest will follow.

Bug: webrtc:6353
Change-Id: I556f6b58f4ca81234ffae3dc6e1319f9c60a76ae
Reviewed-on: https://webrtc-review.googlesource.com/c/117260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26239}
2019-01-14 12:34:51 +00:00
5d7cf3d4eb Fix RTC_CHECK in neteq_rtp_fuzzer
The change in
https://webrtc-review.googlesource.com/c/116683 made the fuzzer crash
at startup.

Bug: chromium:921050, webrtc:10185
Change-Id: Ie3eb26e12b4ae9b29c1c424af0d3eb287b5f1a73
Reviewed-on: https://webrtc-review.googlesource.com/c/117261
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26238}
2019-01-14 12:24:02 +00:00
1a86b78180 Delete StreamInterface methods GetPosition, SetPosition and Rewind
Keep methods on subclasses where they are used: FifoBuffer and
MemoryStream. Also FileStream gets to keep SetPosition, because it's
used by a downstream subclass.

Bug: webrtc:6424
Change-Id: If2a00855aba7c2c9dc0909cda7c8a8ef00e0b9af
Reviewed-on: https://webrtc-review.googlesource.com/c/116487
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26237}
2019-01-14 12:22:22 +00:00
eb0de2284e Remove explicit -fsanitize=memory from BUILD.gn.
There is no need to explicitly do this, because when the GN
argument is_msan is True, the -fsanitize=memory is added by the
toolchain.

Bug: None
Change-Id: Id21dbd56df65636ca038e1abccaada0f44abfbb0
Reviewed-on: https://webrtc-review.googlesource.com/c/116992
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26236}
2019-01-14 11:26:27 +00:00
dc71420e78 Remove deprecated code from logging.
Bug: webrtc:10198
Change-Id: I00c74751c5c71e515c3208a558677215ac547b78
Reviewed-on: https://webrtc-review.googlesource.com/c/116994
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26235}
2019-01-14 11:02:49 +00:00
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