Commit Graph

23774 Commits

Author SHA1 Message Date
5292654d7b Reland "Fix a bug in barcode_decoder.py"
This is a reland of 5c2de6b3ce079cff52c411a2c02ce6553a38dc79

Original change's description:
> Fix a bug in barcode_decoder.py
>
> When converting from a .y4m file, it's illegal to pass a video_size
> option since the resolution is already contained in the .y4m file.
>
> Bug: webrtc:9642
> Change-Id: Iee7d2ba1332c45a1669af0fba43b0c3e7ce5846b
> Reviewed-on: https://webrtc-review.googlesource.com/95949
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24431}

Bug: webrtc:9642
Change-Id: Iea6aad249839f9b1dad830bdf194cef2cc7dcfa6
Reviewed-on: https://webrtc-review.googlesource.com/97441
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24542}
2018-09-04 07:39:13 +00:00
a421775a6d Delete class EventTimerWrapper.
Only user, iSACTest, refactored to use a sleep instead.

Bug: webrtc:3380
Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
Reviewed-on: https://webrtc-review.googlesource.com/96802
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24541}
2018-09-04 06:53:28 +00:00
3613fef7a2 Remove MediaOptimization::Reset.
It is called right after construction, so move the needed
implementation into the MediaOptimization constructor instead.

Bug: webrtc:9711
Change-Id: Ibca35670bf45a85538c34c8ead58ba855acc6b96
Reviewed-on: https://webrtc-review.googlesource.com/97325
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24540}
2018-09-03 15:56:55 +00:00
4e5af96606 Include i420 buffers in Obj-C framework again.
These headers was lost in the cleanup CL for the Obj-C directories. This
puts them back in the framework headers.

Note that since the protocol and interface was split into two different
headers, and all public framework headers are put into a flat directory
structure, I had to rename the implementation files so they would not collide
in the framework header directory.

Bug: webrtc:9701
Change-Id: I42d4c1e02bdfa4e114575f527c4c42a19be8fb52
Reviewed-on: https://webrtc-review.googlesource.com/97330
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24539}
2018-09-03 15:06:18 +00:00
ec76466da2 Adds % unit to float field trial parser.
This can improve readability in some situations and prevents the need
to define a parameter as some_value_in_percent.

Bug: webrtc:9510
Change-Id: I0959d2b6c463f1bc1cea8e66f0bd5b56380b8c03
Reviewed-on: https://webrtc-review.googlesource.com/97302
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24538}
2018-09-03 15:02:58 +00:00
5dd6167908 Echo metric support for the APM-QA.
This was done by
* adding an EchoMetric class to EvaluationScore
* passing an echo metric binary path from the cmd arguments to the
  EvaluationScoreWorkerFactory
* passing the render input filepath to the Evaluator.

The echo score is supposed to be computed by the provided binary. It
should print the echo score in [0.0, 1.0] to a text file. It should
satisfy the cmd flags in its invocation in EchoMetric._Run()


Bug: webrtc:7494
Change-Id: I397013d6ed17659ea01d0623d98a14d4fcdcc161
Reviewed-on: https://webrtc-review.googlesource.com/97022
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24537}
2018-09-03 14:54:23 +00:00
5438bce467 Add field trial for disabling FrameDropper.
Bug: webrtc:9711
Change-Id: Iaa68fa4de589c05cf6b8cab87bf00ad1f3c565f9
Reviewed-on: https://webrtc-review.googlesource.com/97327
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24536}
2018-09-03 14:39:18 +00:00
260182d9f3 Remove unused MockFrameDropper and make FrameDropper non-virtual.
Bug: webrtc:9711
Change-Id: I962039c3ebea1a9445ab3a43071279c4ce8a55cf
Reviewed-on: https://webrtc-review.googlesource.com/97326
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24535}
2018-09-03 14:19:17 +00:00
fd5770df4e Remove obsolete field trial from the tests
Bug: webrtc:8968
Change-Id: I78f5cca98a469dcfbbecba7a16d31e5aac500fc9
Reviewed-on: https://webrtc-review.googlesource.com/97332
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24534}
2018-09-03 14:05:04 +00:00
1beef1a97a Delete VideoSendStream::EnableEncodedFrameRecording.
Use in VideoQualityTest replaced by creating a wrapper for the encoder.

Bug: None
Change-Id: I5c5519e147ca7ddb97696b0d6958a8a1f5cc6e83
Reviewed-on: https://webrtc-review.googlesource.com/94152
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24533}
2018-09-03 13:06:32 +00:00
3bc0166a4e getStats: add kind alias for mediaType
see https://github.com/w3c/webrtc-stats/issues/301

IDL: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats
Change-Id: I7da443bd1cbf07c9a3118ac04329db28b3543b3f

BUG=webrtc:9674

Change-Id: I7da443bd1cbf07c9a3118ac04329db28b3543b3f
Reviewed-on: https://webrtc-review.googlesource.com/96420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24532}
2018-09-03 11:49:30 +00:00
55de08e7ef Restructure neteq_rtpplay into a library with small executable wrapper.
Most of the code in neteq_rtpplay is moved into a factory class for
NetEqTest. The factory method takes the same argc and argv arguments as
neteq_rtpplay.
This CL also adds a small public API for neteq_test to allow easy
integration into external software.

Bug: webrtc:9667
Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d
Reviewed-on: https://webrtc-review.googlesource.com/96100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 10:42:40 +00:00
88c1a9ecbc Adds infinite addition and subtraction to time units.
This prepares for allowing use making arithmetic operators constexpr.

This also makes it easier to use for comparisons with offsets.
Now a > b + 10 ms works even if b is infinite.

Bug: webrtc:9574
Change-Id: Ie36092b72c2ec0f0c541641199a39155f5a796f3
Reviewed-on: https://webrtc-review.googlesource.com/96820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24530}
2018-09-03 09:12:10 +00:00
16e27a1dc5 Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
Original cl: https://webrtc-review.googlesource.com/95721

Bug: webrtc:9239
Change-Id: I7eac85839182bbcecd0d9bd71ae26f6a1c516df4
Reviewed-on: https://webrtc-review.googlesource.com/96401
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24529}
2018-09-03 09:00:01 +00:00
cd87e014f3 Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a

Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> 
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
> 
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}

Bug: webrtc:3380
Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
Reviewed-on: https://webrtc-review.googlesource.com/97320
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24528}
2018-09-03 08:58:11 +00:00
fea4637cfe Adds check for unused field trial parameters.
This adds a dcheck to detect if a FieldTrialParameter has been created
but not used in parsing a field trial. This is an easy mistake to make
and cause extra work debugging why nothing happens.

Also improving the ergonomics of using the parameter and optional
classes. Making it easier to use them as drop in replacements for their
underlying classes. In particular, the optional parameter class
implements and interface more similar to the optional class.

Bug: webrtc:9510
Change-Id: I5a12dd66396fa4cac9c9cf517172ae2f06984060
Reviewed-on: https://webrtc-review.googlesource.com/96761
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24527}
2018-09-03 08:52:51 +00:00
88e1848fd5 Add a StringBuilder class.
String builder is similar to SimpleStringBuilder, but the difference is
that StringBuilder is built around a std::string instance and supports
dynamic resizing.

Change-Id: I874d22e69e639ff9ef3d5929366f4ba71c545787
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/58980
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24526}
2018-09-03 08:48:42 +00:00
3611afc1c1 Roll chromium_revision bbc67a1bd5..4d01f290d4 (587546:588344)
Manual changes:
* Updated generated list of android deps.
* Didn't roll //build: needs resolution to bugs.webrtc.org/9706 first
* Didn't roll tools to head since current head needs to be in sync with //build
  (because of a clang roll).

Change log: bbc67a1bd5..4d01f290d4
Full diff: bbc67a1bd5..4d01f290d4

Changed dependencies:
* src/base: bc614f359d..89b059cf45
* src/ios: 577dec8385..09d71f227a
* src/testing: 6ddd98a72a..a9ff7768f7
* src/third_party: 1a16d6dec5..d461439d92
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d539d93822..5167fb3f66
* src/third_party/depot_tools: cb32668137..e323bd9d22
* src/third_party/freetype/src: 96b5e50090..2c8e6279a7
* src/third_party/libFuzzer/src: 658ff786a2..669ac87aa7
* src/third_party/libvpx/source/libvpx: 0bfab06084..753fd86e86
* src/tools: d59f2cb6e5..cd9a1568ce
DEPS diff: bbc67a1bd5..4d01f290d4/DEPS

No Clang change.

TBR=phoglund@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: If12bc40c5a1c4d6c2bad2882767fd0ee5ad1b271
Reviewed-on: https://webrtc-review.googlesource.com/97321
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24525}
2018-09-03 08:31:28 +00:00
5aec1f68f1 Remove clang:find_bad_constructs suppression from modules/video_capture.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I71350feb45c23d5a0e17e4828c514ce1c353f6a7
Reviewed-on: https://webrtc-review.googlesource.com/96700
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24524}
2018-09-03 07:33:11 +00:00
bb095aa99b Allow send bitrate < start bitrate in RampUpTest.
Primarily, this is intended to reduce flakyness of
RampUpTest.AudioTransportSequenceNumber. We shouldn't expect audio
send rate >= 300 kbps at all time in these tests. And in general, if
it's at all relevant to test that bitrate doesn't drop below the start
bitrate, a perf test isn't the right place for that.

A run of

./third_party/gtest-parallel/gtest-parallel  -r 1000 -w 1000 \
   --gtest_filter=RampUpTest.AudioTransportSequenceNumber \
   out/Release/webrtc_perf_tests

passes when I ran it locally after this change, but fails around 4 out
of 1000 times before the change.

Bug: webrtc:8878
Change-Id: I08614ce5683c9ba6fe4b72bfde83e6a81445a59b
Reviewed-on: https://webrtc-review.googlesource.com/96900
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24523}
2018-09-03 07:28:39 +00:00
689b5874d4 Use monotonic clock for PhysicalSocketServer timeouts.
Bug: webrtc:9684
Change-Id: Ia50e2d8f8100364ca8047b9b6cf55674206d8d8b
Reviewed-on: https://webrtc-review.googlesource.com/96680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24522}
2018-09-03 07:05:56 +00:00
1606d5eee8 Roll chromium_revision c1d4701..bbc67a1bd5 (585833:587546)
Change log: c1d4701..bbc67a1bd5
Full diff: c1d4701..bbc67a1bd5

Manual changes:
* Added proguard

Changed dependencies:
* src/build: 57d26a0c82..6a5f1f3698
* src/testing: 9369f699ce..6ddd98a72a
* src/third_party: c1acec6af3..1a16d6dec5
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/01e8e625ad..7f4f41fa81
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1e44d06625..d539d93822
* src/third_party/depot_tools: 7b7eb8800b..cb32668137
* src/third_party/googletest/src: d526632675..2e68926a9d
* src/third_party/libjpeg_turbo: a1750dbc79..61a2bbaa9a
* src/third_party/libvpx/source/libvpx: dbcb89be24..0bfab06084
* src/tools: 86ee5d2701..d59f2cb6e5
DEPS diff: c1d4701..bbc67a1bd5/DEPS

No update to Clang.

TBR=phoglund@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I10a4ecab485742818363b76473f9ea45c12aec82
Reviewed-on: https://webrtc-review.googlesource.com/96841
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24521}
2018-09-01 13:54:15 +00:00
e80e0132c1 Remove MSVC debug bots from CQ.
We can no longer keep these up because third_party/protobuf breaks MSVC
after https://chromium-review.googlesource.com/c/chromium/src/+/1081411

No-try: true
Bug: webrtc:9695
Change-Id: I01671d78acc9e89035d3b48973e90bd81776948a
Reviewed-on: https://webrtc-review.googlesource.com/97040
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24520}
2018-09-01 09:20:03 +00:00
8dda03515b ScreenCapturerMac: destroy the streams and remove the DisplayStreamManager
Thanks to the CL https://webrtc-review.googlesource.com/c/src/+/83822
there is now no need to destroy the stream asynchronously.
But the CL above introduced a leak, the streams were stopped but
not destroyed. This CL essentially fixes the leak, it is now safe
to destroy the stream right after being stopped as everything happen
in the same capture thread.

Bug: chromium:851883
Change-Id: I4bf7409246f3957d90040d0d8cf09e98f28d6559
Reviewed-on: https://webrtc-review.googlesource.com/96621
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24519}
2018-08-31 23:49:29 +00:00
9a4fd9bf74 Use AsyncInvoker in DtmfSender instead of MessageHandler
Bug: webrtc:9702
Change-Id: Ib9a9a2cf5bbb7aff24e6690deca51a021961ead3
Reviewed-on: https://webrtc-review.googlesource.com/97182
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24518}
2018-08-31 22:49:26 +00:00
044a04d8b5 Use AsyncInvoker in DataChannel instead of MessageHandler
Bug: webrtc:9702
Change-Id: I76a6a97f792be632c1c2f4f5cbbd26a7ec243006
Reviewed-on: https://webrtc-review.googlesource.com/97183
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24517}
2018-08-31 22:48:20 +00:00
d25828a0bf Use AsyncInvoker in JsepTransportController instead of MessageHandler
Bug: webrtc:9702
Change-Id: I9171d6e7f16fe50be1c2b139bf7dd1d097000791
Reviewed-on: https://webrtc-review.googlesource.com/97181
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24516}
2018-08-31 22:12:45 +00:00
bb19276a32 Use AsyncInvoker in PeerConnection instead of MessageHandler
Bug: webrtc:9702
Change-Id: I89d66d1165a096601aed37b8febad60620073899
Reviewed-on: https://webrtc-review.googlesource.com/97180
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24515}
2018-08-31 20:10:30 +00:00
ed1f75ab6d Removes redundant starting rate.
Sine a starting rate field was added to the constraints struct. Having
it in the initial config separately is reduntant. To simplify the code,
the extra field is removed. This is a follow up on:
https://webrtc-review.googlesource.com/c/src/+/92624

Bug: webrtc:9586
Change-Id: I9b01b16b2fc4b8479e83b7e998308be2295e0325
Reviewed-on: https://webrtc-review.googlesource.com/96801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24514}
2018-08-31 17:36:21 +00:00
836a7a2e4d AEC3: option for using the stationarity estimator at render from the beginning of the call
Bug: webrtc:9697
Change-Id: I2427e9e62505d27b0942fd6b2e38eee6d720f4f3
Reviewed-on: https://webrtc-review.googlesource.com/97081
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24513}
2018-08-31 17:07:02 +00:00
2165233874 Uninline non-trivial AudioOptions functions
reimplement ToString using rtc::SimpleStringBuilder instead of std::ostringstream
Side effect: ToString converts booleans as 0/1 instead of false/true.

Bug: None
Change-Id: I8a57d208b016d3af5a09f7dc2e2ec4e5634446fa
Reviewed-on: https://webrtc-review.googlesource.com/95080
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24512}
2018-08-31 16:25:48 +00:00
a10d164b4a Implement periodic cancelable task for task queue
using shared pointer to boolean flag.

Bug: None
Change-Id: I9d7ad7d7b187fefa7daa0247a1379e1ddd7e2b24
Reviewed-on: https://webrtc-review.googlesource.com/96300
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24511}
2018-08-31 16:10:20 +00:00
d05916dccf Bump xcode versions for WebRTC bots.
The build toolchain now requires xcode 10, so we have to roll.

Bug: None
Change-Id: Iafec62e7927ca8a81117710d09e2c42bcf18c0d1
Reviewed-on: https://webrtc-review.googlesource.com/97060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24510}
2018-08-31 15:32:10 +00:00
d4161a3c9d Moving LappedTransform, Blocker and AudioRingBuffer.
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.

Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
2018-08-31 15:27:50 +00:00
8a3c166fff Cleanup RtpPacketizerVP8 tests
Remove partition support for test helper and from tests.
Merge Init function into constructor
Replace extra macroses in favor of Bit helper function
Replace extra members in favor of local variables
Remove fixture

Bug: None
Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea
Reviewed-on: https://webrtc-review.googlesource.com/96940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24508}
2018-08-31 08:13:45 +00:00
d3b8c63b58 Reland "Add spatial index to EncodedImage."
This is a reland of da0898dfae3b0a013ca8ad3828e9adfdc749748d

Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}

Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
2018-08-31 07:35:52 +00:00
240215431e AEC3: Parametrize the shadow filter output usage
This CL introduces the ability to control the usage of the shadow filter
output in the echo canceller output.

Bug: webrtc:9694,chromium:879451
Change-Id: I01f90de60de1799b32892051c176bda5e1a8d33e
Reviewed-on: https://webrtc-review.googlesource.com/97020
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24506}
2018-08-31 06:51:16 +00:00
558b93b3e9 Add the multicast DNS message format.
This CL adds the utilities to generate and parse mDNS messages (RFC 1035
and RFC 6762).

TBR=phoglund@webrtc.org

Bug: webrtc:9605
Change-Id: Id6121c17926887cd3a41a2dfc829462fd15f3a4c
Reviewed-on: https://webrtc-review.googlesource.com/93241
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24505}
2018-08-31 00:02:44 +00:00
cc22f51988 Removing the intelligibility enhancer.
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).

Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
2018-08-30 21:29:57 +00:00
fc173d00ec Add no_size_t_to_int_warning suppression to webrtc.
Current set of warnings that need fixing before this warning can
be enabled is here -> https://pastebin.com/raw/jTddgPzP

BUG=chromium:588506, webrtc:9451

Change-Id: Id7896cf48c7231b2ee28dde378ff3ce17da73c2b
Reviewed-on: https://webrtc-review.googlesource.com/96623
Commit-Queue: Will Harris <wfh@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24503}
2018-08-30 17:45:54 +00:00
260b4151c8 Revert "Reland "Optimize execution time of RTPSender::UpdateDelayStatistics""
This reverts commit 7bcd2a98be3fa8c246866d6b343c7f94752977b3.

Reason for revert: peerconnection_unittests fails on downstream test runner.

Original change's description:
> Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
> 
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
> 
> Bug: webrtc:9439
> Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
> Reviewed-on: https://webrtc-review.googlesource.com/96641
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24501}

TBR=terelius@webrtc.org,asapersson@webrtc.org,philipel@webrtc.org

Change-Id: Ia48444d2a7647cf826ef93b4720f6d7ff9a712c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/96960
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24502}
2018-08-30 15:24:47 +00:00
7bcd2a98be Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.

Bug: webrtc:9439
Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
Reviewed-on: https://webrtc-review.googlesource.com/96641
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24501}
2018-08-30 14:42:02 +00:00
c714b6e713 Adds TaskQueue congestion controller tests in VideoSendStreamTest.
Bug: webrtc:8415
Change-Id: If49d228cc9440e19fbf73c771ceece86b444c4c0
Reviewed-on: https://webrtc-review.googlesource.com/92625
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24500}
2018-08-30 14:14:42 +00:00
ba3decfe00 Adds support for frame rate control in FrameGeneratorCapturer.
Bug: webrtc:9510
Change-Id: Idb062978a856ead994fce7e72402effcd8c719cf
Reviewed-on: https://webrtc-review.googlesource.com/95148
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24499}
2018-08-30 13:52:46 +00:00
34ee419a23 Fixes breaking bug in feedback based GoogCC.
This CL fixes a bug in the feedback based GoogCC where packets lost
was swapped with expected packets received. Since this version of
GoogCC isn't yet used this wasn't discovered. There was also a lack
of unit test coverage. To ensure reasonable behavior, unit tests was
added.

Unit tests was also converted from relevant unit tests on send side
congestion controller for the regular GoogCC controller.

Bug: webrtc:9586
Change-Id: I83c40ff4766104820cb72ec1e8b95c5782def19a
Reviewed-on: https://webrtc-review.googlesource.com/59401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24498}
2018-08-30 13:22:51 +00:00
498aceb3d8 Roll chromium_revision 33a17747bb2..c1d47013a1 (585798:585833)
Manual changes:
* Added dagger libs

Change log: 33a17747bb..c1d47013a1
Full diff: 33a17747bb..c1d47013a1

* src/base: 7cf937ddfd..bc614f359d
* src/build: 53a2dfe471..57d26a0c82
* src/ios: 34ebe8d53e..577dec8385
* src/third_party: 951cbbaec1..c1acec6a
* src/tools: 0181226..86ee5d2701

No update to Clang.

TBR=phoglund@webrtc.org

CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Bug: None
Change-Id: I53a1eb04ebb879cb86095c416dfd1f28ce7b7e36
Reviewed-on: https://webrtc-review.googlesource.com/96821
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24497}
2018-08-30 13:06:11 +00:00
6be91eb2f8 Revert "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
This reverts commit 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a.

Reason for revert: Makes the perf test RampUpTest.AudioTransportSequenceNumber fail on windows, almost every time.

Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> 
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
> 
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}

TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org

Change-Id: I8867a22d695494bd5abfda6a97f0719cb3ff3d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/96840
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24496}
2018-08-30 12:59:13 +00:00
b0d4b415cc Use a lock to protect members accessed by RtpVideoStreamReceiver::GetSyncInfo()
Proper synchronization was overlooked in
https://webrtc-review.googlesource.com/93261

Bug: chromium:878319, webrtc:7135
Change-Id: Ifc850c4d67a4e9dd2660dab9b6da67258338553e
Reviewed-on: https://webrtc-review.googlesource.com/96461
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24495}
2018-08-30 12:33:43 +00:00
5304a32a94 Delete StreamStatistician::IsRetransmitOfOldPacket
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.

Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
2018-08-30 11:00:13 +00:00
7bca8ca4e2 Obj-C SDK Cleanup
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
2018-08-30 10:42:41 +00:00