Commit Graph

22787 Commits

Author SHA1 Message Date
b90e63c620 Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet
Bug: webrtc:9370
Change-Id: I59606ef6ea9bbf26de844a2fd3f597856271a86a
Reviewed-on: https://webrtc-review.googlesource.com/81700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23555}
2018-06-08 14:58:18 +00:00
ec9c745228 Adds support for new Windows ADM with limited API support.
Summary of what this CL does:

Existing users can keep using the old ADM for Windows as before.

A new ADM for Windows is created and a dedicated factory method is used
to create it. The old way (using AudioDeviceImpl) is not utilized.

The new ADM is based on a structure where most of the "action" takes
place in new AudioInput/AudioOutput implementations. This is inline
with our mobile platforms and also makes it easier to break out common
parts into a base class.

The AudioDevice unittest has always mainly focused on the "Start/Stop"-
parts of the ADM and not the complete ADM interface. This new ADM supports
all tests in AudioDeviceTest and is therefore tested in combination with
the old version. A value-parametrized test us added for Windows builds.

Improved readability, threading model and makes the code easier to maintain.

Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility.

Bug: webrtc:9265
Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d
Reviewed-on: https://webrtc-review.googlesource.com/78060
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 14:44:38 +00:00
488eb98616 Setting resolution alignment to 4 on iOS.
Bug: webrtc:9381
Change-Id: I6fb6cc6ffa197ca581462e308a857ac38e10b9a1
Reviewed-on: https://webrtc-review.googlesource.com/82162
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23553}
2018-06-08 14:17:07 +00:00
443e71f528 Revert "Disabling VeryLowBitrateVP9 to unblock roll."
This reverts commit 16e28d143a32ff3552efe0a014178f68006812b8.

Reason for revert: Fix has supposedly landed upstream.

Original change's description:
> Disabling VeryLowBitrateVP9 to unblock roll.
> 
> This should be re-enabled very soon since the libvpx thinks this
> is fixed upstream and is only waiting for merge.
> 
> TBR=marpan@google.com
> 
> Bug: webrtc:9292
> Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
> Reviewed-on: https://webrtc-review.googlesource.com/81660
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23525}

TBR=phoglund@webrtc.org,marpan@google.com

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

Bug: webrtc:9292
Change-Id: I995953070536e8ee3540e7c30bc11dc1200e0463
Reviewed-on: https://webrtc-review.googlesource.com/82200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23552}
2018-06-08 13:55:25 +00:00
2d61162831 Remove stringstreams from media/sctp/
Bug: webrtc:8982
Change-Id: I0d92f56b628e7cd50e7c853d3bfe0049dcf71425
Reviewed-on: https://webrtc-review.googlesource.com/76563
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23551}
2018-06-08 13:51:51 +00:00
25b41f8c11 remove unused stringstream import
No-Try: true
Bug: webrtc:8982
Change-Id: I24537a3d4fab2d0caa4e62ed791c9939be8e4567
Reviewed-on: https://webrtc-review.googlesource.com/77120
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23550}
2018-06-08 13:03:34 +00:00
0428d37305 Roll chromium_revision f9a24d5d6f..e9fcce491e (564886:565609)
Change log: f9a24d5d6f..e9fcce491e
Full diff: f9a24d5d6f..e9fcce491e

Roll chromium third_party addc496eeb..b1a95b2d51
Change log: addc496eeb..b1a95b2d51

Changed dependencies:
* src/base: 769dcfb902..6d59c6b0f4
* src/build: 2574ebeaaf..472d5df73f
* src/ios: 044af8282c..2c466eefd8
* src/testing: 5ab6ab6e1c..4162fb6122
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5601bdac1a..5267ef7b4a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/01f608bbcf..c9718952e5
* src/third_party/depot_tools: e89dcf7dee..8e6f58c7e6
* src/third_party/libvpx/source/libvpx: 3f7e6cc020..87386826a9
* src/tools: 8a16887980..1702b5a977
DEPS diff: f9a24d5d6f..e9fcce491e/DEPS

No update to Clang.

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

Change-Id: I28be447ede8150e4030611bdf3d1e4dda49ea32a
Reviewed-on: https://webrtc-review.googlesource.com/82120
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23549}
2018-06-08 12:11:51 +00:00
6aa415e5e3 Add full stack test with link capacity of 150 kbit/s.
Useful, because this is lower then the video pipeline's default start
bitrate of 300 kbit/s.

Bug: webrtc:9176
Change-Id: Iff9fc883df76f4b9265d89bcbd97c23ea45c3a51
Reviewed-on: https://webrtc-review.googlesource.com/80841
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23548}
2018-06-08 11:33:30 +00:00
350531e2a3 Revert "Move class VideoCodec from common_types.h to its own api header file."
This reverts commit efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd.

Reason for revert: probably breaks downstream test

Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
> 
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}

TBR=danilchap@webrtc.org,brandtr@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: Id8bd37c79c2f8d09a4d88368765230103f1db2c8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7660
Reviewed-on: https://webrtc-review.googlesource.com/82101
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23547}
2018-06-08 11:04:23 +00:00
72520531e8 Add R8 to DEPS.
Chromium has switched to using the r8 dexer from dx, so we need to
switch as well.

Bug: None
Change-Id: I8c7845bd100ad21c6142a35e7f266a2aa37563b0
Reviewed-on: https://webrtc-review.googlesource.com/82062
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23546}
2018-06-08 10:31:38 +00:00
9edbc27960 Roll chromium_revision 42930fc83a..f9a24d5d6f (564069:564886)
Change log: 42930fc83a..f9a24d5d6f
Full diff: 42930fc83a..f9a24d5d6f

Roll chromium third_party 2f3180d1a4..addc496eeb
Change log: 2f3180d1a4..addc496eeb

Changed dependencies:
* src/base: 5d5b155006..769dcfb902
* src/build: de1de7df37..2574ebeaaf
* src/buildtools: 893eb86b02..6f4dae280c
* src/ios: 78bc41f03c..044af8282c
* src/testing: 36f6919025..5ab6ab6e1c
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/54e864df3f..01f608bbcf
* src/third_party/depot_tools: 518bb6847a..e89dcf7dee
* src/third_party/libvpx/source/libvpx: 2b08f89076..3f7e6cc020
* src/tools: bea2920553..8a16887980
DEPS diff: 42930fc83a..f9a24d5d6f/DEPS

No update to Clang.

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

Change-Id: I22104809b6bec8ab773411983357f2d385332395
Reviewed-on: https://webrtc-review.googlesource.com/82060
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23545}
2018-06-08 07:56:44 +00:00
efc71e565e Move class VideoCodec from common_types.h to its own api header file.
Bug: webrtc:7660
Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
Reviewed-on: https://webrtc-review.googlesource.com/79881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23544}
2018-06-08 07:55:04 +00:00
b616d09e1b Android: Add deprecation annotations to VideoRenderer and I420Frame
Bug: webrtc:9181
Change-Id: Ice569c88ce1a2cda6a314f2889c8b64b55cb8c8f
Reviewed-on: https://webrtc-review.googlesource.com/81746
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23543}
2018-06-08 07:46:24 +00:00
5aba818e45 Remove test AudioCodingModuleTest.TestAPI
Since it isn't being run by the bots, it has bit rotted; when I try to
run it manually, it fails with a long list of error messages:

  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
  >>>   Error Enabling VAD    <<<
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
  >>>   Error Enabling DTX    <<<
  >>>   Error Enabling VAD    <<<
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
  >>>   Error Enabling VAD    <<<
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996
  Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 985

...and so on.

Bug: webrtc:8396
Change-Id: Id8f1e01a751b4bb3527702b7b7a4986ce0abb378
Reviewed-on: https://webrtc-review.googlesource.com/81745
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23542}
2018-06-08 07:45:20 +00:00
5e8fd8ad49 Add simulcastStream output from VideoCodecTestFixture::Config::ToString.
Bug: None
Change-Id: I06c6ac077bb31608b4776e90d548a6e71ca1c252
Reviewed-on: https://webrtc-review.googlesource.com/81186
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23541}
2018-06-08 07:37:50 +00:00
086de82f51 Add bitrate_priority to GetSimulcastConfig call.
Bug: webrtc:9368
Change-Id: I72317493db02835362c0e6127e6e4c25a5709d63
Reviewed-on: https://webrtc-review.googlesource.com/81661
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23540}
2018-06-08 07:04:44 +00:00
e266c80ce3 Guard the android aaudio headers include with the aaudio preprocessor
Bug: webrtc:9365
Change-Id: Iba7ea24abb1b4d617a61079b3ec2ed2a3148c865
Reviewed-on: https://webrtc-review.googlesource.com/81341
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23539}
2018-06-07 20:00:15 +00:00
ed51a6e665 AEC3: Avoid static initializers
Bug: webrtc:9288,chromium:846615
Change-Id: I9df7f07454bdba45181972b7ed3dff77c370abb3
Reviewed-on: https://webrtc-review.googlesource.com/81750
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23538}
2018-06-07 18:13:01 +00:00
05d8ee1b3e AEC3: Delay stabilization after a delay change
This CL ensures that the linear-filter based refined delay is chosen to
match the delay that was detected by the delay estimator during the time
it takes for the linear filter to converge.

Bug: webrtc:9371,chromium:850451
Change-Id: Ib9cf532df0577ceca10a260d9d2deba5306f88bb
Reviewed-on: https://webrtc-review.googlesource.com/81682
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23537}
2018-06-07 14:35:55 +00:00
78ea818864 AEC3: Added filter preprocessing to avoid low frequency artefacts
This filter preprocess the time domain representation of the adaptive
linear filter to avoid low-frequency components causing issues in
the filter analysis.

Bug: webrtc:9343, chromium:848231
Change-Id: I40494959f1b76242a7c9f2a2fc85c2ad4af9e164
Reviewed-on: https://webrtc-review.googlesource.com/79142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23536}
2018-06-07 13:35:40 +00:00
3ecec84c83 Android: Throw exception in CallSessionFileRotatingLogSink if dir is null
This CL throws an IllegalArgumentException in case the dir path argument
is null. This makes the error more clear than crashing in native JNI
code while trying to convert a null string.

Bug: b/106732994
Change-Id: Ib04ebf017c6e33b9896fc1e1db051a853838a7f4
Reviewed-on: https://webrtc-review.googlesource.com/81740
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23535}
2018-06-07 13:16:40 +00:00
f469b63d44 AEC3: Improved anti-aliasing filter for DSF 4
This change contains a new anti-aliasing filter for the delay estimator
for down-sampling factor 4. The new (elliptic) filter has a much wider
main lobe allowing for faster convergence.

Bug: webrtc:9288,chromium:846615
Change-Id: Id109974a59fe6f48c5e0ccc4f4e06c0d94c8bd03
Reviewed-on: https://webrtc-review.googlesource.com/81680
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23534}
2018-06-07 12:21:36 +00:00
7210af206c Removing warning suppression flags from test/.
Bug: webrtc:9251
Change-Id: Ibe3971adcc13d30d4a3360ecfe6d525e02428b28
Reviewed-on: https://webrtc-review.googlesource.com/81188
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23533}
2018-06-07 11:54:56 +00:00
1127fb9856 Re-enabling SanitizerTest.MsanUninitialized.
Using EXPECT_DEATH instead of __msan_set_expect_umr.

Bug: webrtc:9116
Change-Id: I40222e920cd213d3d1366629023d4d1c4fbc35f9
Reviewed-on: https://webrtc-review.googlesource.com/79661
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23532}
2018-06-07 11:39:15 +00:00
b544f6c2f5 Fixing issue where pacer budget increased in congestion.
This fixes an issue where the media budget in the pacer was allowed to
increase more than the process interval when congested.

Bug: webrtc:8415
Change-Id: I79bf965b6a72ed88313074cdae4746fcaff63340
Reviewed-on: https://webrtc-review.googlesource.com/80121
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23531}
2018-06-07 10:13:48 +00:00
6cb74fd77a Remove unused methods in VCMDecoderDataBase.
Bug: none
Change-Id: Ice538b4be577b4a474b9a16bcec4977eb73d22fb
Reviewed-on: https://webrtc-review.googlesource.com/80540
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23530}
2018-06-07 08:46:57 +00:00
97e04884bd Delete unused stats for preferred_bitrate.
Bug: webrtc:8830
Change-Id: Iaa30488255f2e09e269274136d370740cd030902
Reviewed-on: https://webrtc-review.googlesource.com/78880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23529}
2018-06-07 08:11:07 +00:00
34c9f1252a AEC3: Move decimator filters to the new notation
Preparing for changing the filters of the decimator by moving the old
filters to the new zero, pole, gain notation.

Bug: webrtc:9288,chromium:846615
Change-Id: I2b01a2555d34617e0bf251c782703753f72cd56f
Reviewed-on: https://webrtc-review.googlesource.com/81189
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23528}
2018-06-07 08:09:17 +00:00
2b3af2e8be Delete RTP-specific values from the VideoCodecType enum.
Bug: None
Change-Id: Icd6a03f4dc7cfe074ba1e0370ed40938f0f1d7ed
Reviewed-on: https://webrtc-review.googlesource.com/80442
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23527}
2018-06-07 07:49:27 +00:00
81327d54f3 Move stats for delayed frames to renderer from VCMTiming to ReceiveStatisticsProxy.
Bug: none
Change-Id: If62cc40cf00bc4d657a31a89640d03812cff388e
Reviewed-on: https://webrtc-review.googlesource.com/74500
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23526}
2018-06-07 07:39:37 +00:00
16e28d143a Disabling VeryLowBitrateVP9 to unblock roll.
This should be re-enabled very soon since the libvpx thinks this
is fixed upstream and is only waiting for merge.

TBR=marpan@google.com

Bug: webrtc:9292
Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
Reviewed-on: https://webrtc-review.googlesource.com/81660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23525}
2018-06-07 07:34:27 +00:00
a3e69e6c74 Add min_bitrate_bps to RTCRtpEncodingParameters.
This is an ObjC followup to https://webrtc-review.googlesource.com/c/src/+/78741.

This CL only adds the field to the API, but does not wire it up.

Bug: webrtc:9341
Change-Id: Id6b1ac681324120bc90158029da7a80bf99aa512
Reviewed-on: https://webrtc-review.googlesource.com/81182
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23524}
2018-06-07 07:26:07 +00:00
1659e97767 Use DecimalFormat instead of String.format.
Speculative fix to a problem where String.format crashes with
"java.lang.NullPointerException at Formatter$FormatSpecifier.addZeros :
Attempt to get length of null array" on some devices.

Bug: b/80240768
Change-Id: I8e67b7107a37ad7d6f978b9de368f14d37efecb2
Reviewed-on: https://webrtc-review.googlesource.com/80883
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23523}
2018-06-05 14:28:04 +00:00
5daaf7dbc6 Support cropping and rotation override in Metal renderers.
Bug: webrtc:9301
Change-Id: Ic761f0fd6ad6fee74021b84903f1653878453533
Reviewed-on: https://webrtc-review.googlesource.com/80460
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23522}
2018-06-05 14:19:14 +00:00
e4e1ce903b Android: Remove deprecated CapturerObserver methods
Bug: webrtc:9181
Change-Id: Ibf558102f41bf9fa3fcfa2b4bab7c7675e889bfc
Reviewed-on: https://webrtc-review.googlesource.com/77682
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23521}
2018-06-05 14:03:56 +00:00
b6c462d4e4 Cleanup webrtc:: namespace from leaked TimingFrameFlags
Bug: webrtc:9351
Change-Id: Ifbc0a522bf13ab62a2e490b9f129eacfabe7796f
Reviewed-on: https://webrtc-review.googlesource.com/80961
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23520}
2018-06-05 13:52:04 +00:00
5352ad7928 Remove MIPS MB config since we don't need it anymore.
Bug: webrc:9353
Change-Id: I1205ff86597c149a574e46219245cfe0f741875c
Reviewed-on: https://webrtc-review.googlesource.com/80885
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23519}
2018-06-05 12:35:20 +00:00
4c6390aabc Remove deprecated RtpSenderInterface::GetParameters() const method
Bug: webrtc:7580
Change-Id: If6e44509256508a23c5903324f29560157974986
Reviewed-on: https://webrtc-review.googlesource.com/80962
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23518}
2018-06-05 11:43:20 +00:00
45a57fda24 Remove unused include from FrameBuffer2.
Bug: None
Change-Id: I766b430beb4f5ba35519931fbff19261a462f2c2
Reviewed-on: https://webrtc-review.googlesource.com/81184
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23517}
2018-06-05 11:33:20 +00:00
dac422f5d3 Fixing gtest_parallel_wrapper_test on Windows.
Using tempfile is probably overkill in this case, but it is good to
have a meaningful path printed out in case of error (instead of
something like "/tmp" and then a Windows path).

Bug: None
Change-Id: I90b939d7b2a082f4c04f995b602942efe1e671bc
Reviewed-on: https://webrtc-review.googlesource.com/81180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23516}
2018-06-05 11:29:30 +00:00
3ea3e300dc Fixing some SIGFPEs that are making my tests crash
Bug: none
Change-Id: Ib538e4f131a2c05b9b832bc8235f4f0bb35d04c0
Reviewed-on: https://webrtc-review.googlesource.com/74622
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23515}
2018-06-05 10:03:48 +00:00
3a79a9a290 Remove deprecated API methods in video pipeline
Bug: none
Change-Id: I3c3d493f9e14a93868c86fa94ef7269126bd9877
Reviewed-on: https://webrtc-review.googlesource.com/80482
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23514}
2018-06-05 08:26:05 +00:00
ab55c34ef1 Remove warning suppression flags from sdk/android.
Bug: webrtc:9251
Change-Id: Iafd10a09dd34dfdd590711bb5c6a36d08298d1e7
Reviewed-on: https://webrtc-review.googlesource.com/79882
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23513}
2018-06-04 20:08:40 +00:00
267287407a AEC3: Set default down sampling factor to 4
This change sets the default down sampling factor of the delay estimator
4. Down sampling factor 8 is still problematic for some signals.

Bug: webrtc:9288,chromium:846615
Change-Id: I125371f3caf4f660ea2a386d2b89d7a488dcf802
Reviewed-on: https://webrtc-review.googlesource.com/80965
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23512}
2018-06-04 19:04:10 +00:00
27300c3546 Allow 3 encoder threads in libvpx for HD on > 6 core cpus
Bug: webrtc:4172
Change-Id: I50446779403eff0fe2e840afc6cfab9f8a310b1a
Reviewed-on: https://webrtc-review.googlesource.com/77981
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23511}
2018-06-04 16:48:09 +00:00
ac1501acd9 Throw exception in Java if log tag or message is null
Change-Id: If1d2cefcdf5441381bd6c06416cab24883d31481
Bug: webrtc:9357
Reviewed-on: https://webrtc-review.googlesource.com/80963
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23510}
2018-06-04 14:58:09 +00:00
a24493e048 Roll chromium_revision cf1645bec7..42930fc83a (563963:564069)
Change log: cf1645bec7..42930fc83a
Full diff: cf1645bec7..42930fc83a

Roll chromium third_party 1b4d922d36..2f3180d1a4
Change log: 1b4d922d36..2f3180d1a4

Changed dependencies:
* src/base: 57de7ba6db..5d5b155006
* src/build: ea9199f1f4..de1de7df37
* src/ios: c790da9b2a..78bc41f03c
* src/testing: e10eefe10d..36f6919025
* src/third_party/depot_tools: 88087bad47..518bb6847a
* src/tools: 67b206d389..bea2920553
DEPS diff: cf1645bec7..42930fc83a/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
NO_AUTOIMPORT_DEPS_CHECK=true

Change-Id: I632df5f339c386141eec8731e69deb1d833800b0
Reviewed-on: https://webrtc-review.googlesource.com/80920
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23509}
2018-06-04 13:06:07 +00:00
adb4841173 Remove explicit locking using av_lockmgr_register
av_lockmgr_register is deprecated and no-op since
a04c2c707d

Bug: webrtc:8745
Change-Id: I284c9a6edf88a584c3a5cb5dfae4ccf1be1f8851
Reviewed-on: https://webrtc-review.googlesource.com/39503
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23508}
2018-06-04 12:17:07 +00:00
520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00
b18931d30c Updating ffmpeg deprecated functions TODO.
webrtc:8745 is closed and it was about unblocking the Chromium roll,
webrtc:9352 is the new bug to keep track of the removal of ffmpeg
deprecated functions.

Bug: webrtc:9352
Change-Id: I2818dba804f3d611d4df80559a635e7cf1ee5338
No-Try: True
TBR: phoglund@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/80882
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23506}
2018-06-04 11:48:47 +00:00