Commit Graph

24607 Commits

Author SHA1 Message Date
8c27ccac75 Promotoing webrtc::CryptoOptions to RTCConfiguration.
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.

To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.

Got LGTM offline from Sami, adding him to TBR if he has any further comments.

TBR=sakal@webrtc.org

Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
2018-10-25 17:59:48 +00:00
78410ad413 Fixes use after free error when setting a new FrameEncryptor on ChannelSend.
This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be  deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.

To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.

ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.

Added Seth as TBR as this only introduces mocks.

TBR=shampson@webrtc.org

Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}
2018-10-25 17:36:57 +00:00
f26e290e33 fuchsia: Stub out timing and memory functions
This functionality isn't (currently) available on Fuchsia from the OS.

Bug: chromium:808287
Change-Id: If017bc762448c437b74cb03587ba35da5d131c75
Reviewed-on: https://webrtc-review.googlesource.com/c/107760
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Scott Graham <scottmg@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25373}
2018-10-25 17:13:00 +00:00
9c8ae4b9a1 Disable probe delay warning in release builds.
This log is triggering many times a second for Chrome Remote Desktop on some
browsers. This CL just turns it off for release builds to avoid log files
filling up users' disks until we figure out what's going on.

Bug: chromium:888038
Change-Id: Ibbe9d47295b3633314feb28e155e3f59b878dbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/107688
Commit-Queue: Jamie Walch <jamiewalch@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25372}
2018-10-25 16:00:16 +00:00
6c6c9df99d Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
Underscore methods in the middle of classes is against the chromium style guide
this change is part of a long series of changes to refactor crypto code in
WebRTC to conform to the chromium standard better.

1. ssl_cert() -> GetSSLCertificate()
2. ssl_cert_chain() -> GetSSLCertificateChain()
3. Small tidying up in rtccertificategenerator.cc

Bug: webrtc:9860
Change-Id: I670f76e31d6d4f873034edb72d958b3c227379cb
Reviewed-on: https://webrtc-review.googlesource.com/c/107802
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25371}
2018-10-25 15:52:06 +00:00
359d60a594 Adds target rate to audio send stream stats.
Bug: webrtc:9510
Change-Id: I8bd74fc115e3006f477b289edc58fa1f9d7b6bc6
Reviewed-on: https://webrtc-review.googlesource.com/c/107652
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25370}
2018-10-25 15:12:36 +00:00
57ba7e1276 Normalize baseline in network delay plot to RTT/2.
Bug: None
Change-Id: I0d4266216adf9283ceb335281f9332f66a04324e
Reviewed-on: https://webrtc-review.googlesource.com/c/107648
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25369}
2018-10-25 14:19:55 +00:00
039743e066 Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc
Reviewed-on: https://webrtc-review.googlesource.com/c/107716
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-25 14:13:44 +00:00
e2754c95c5 Fixes bug in AudioPriorityBitrateAllocationStrategy field trial.
Previously the rate limits weren't properly applied. This is fixed by
working on mutable copies of the TrackConfig.

Bug: webrtc:9718
Change-Id: I7438c59efa5d7e70fa3ce5e466e2c53a5a7ea9e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107636
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25367}
2018-10-25 13:59:22 +00:00
c0e4d45ce0 Adds BitrateAllocation struct to OnBitrateUpdated.
This prepares for adding parameters to OnBitrateUpdated. By using a
struct, additional fields doesn't require a change in the signature and
only the obeservers that use the new fields will be affected by the
change.

Bug: webrtc:9718
Change-Id: I7dd6c9577afd77af06da5f56aea312356f80f9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/107727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25366}
2018-10-25 13:51:11 +00:00
4ba6c26623 Delete MessageData when a message is posted to a quitting MessageQueue
Bug: webrtc:9913
Change-Id: Id60f537eb0049995c9f0837e3a03ca3a3dd90577
Reviewed-on: https://webrtc-review.googlesource.com/c/107639
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25365}
2018-10-25 13:36:32 +00:00
9516c38538 [Fuzzer] Check FieldTrial bitmask size at compile time.
Rather fail at compile time than at run-time.

Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
2018-10-25 13:21:31 +00:00
1803bb2470 Fix for clock read race in FakeNetworkPipe.
Bug: none
Change-Id: Id708c532bfc0c9cd696a974d455ff79f25c222fe
Reviewed-on: https://webrtc-review.googlesource.com/c/107880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25363}
2018-10-25 12:34:01 +00:00
3284b61a6c Fix for packet loss tracking in network emulation.
Fake_network_pipe currently only counts losses due to buffer overflow.
Fix by counting all packets marked as lost.

Bug: webrtc:9904
Change-Id: I070538b289d925c650d8abca1644ba015227c2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/107646
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25362}
2018-10-25 12:32:05 +00:00
262047055d Update fuzzer max input length handling
The docs have been updated. max_len is libfuzzer specific, new way is
fuzzer agnostic.

Docs:
https://chromium.googlesource.com/chromium/src/+/master/testing/libfuzzer/getting_started.md#improving-your-fuzz-target

Bug: chromium:895082
Test: flexfec_sender_fuzzer input size still converges at <=200 after running locally for 5-10 minutes.
Change-Id: I7a5ce95cb4d8b8ca461f6e502b81b599daa855f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107883
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25361}
2018-10-25 12:19:18 +00:00
ddc84e9819 Publish function_video_(en|de)coder_factory into api
Bug: None
Change-Id: Ibdae580c085cfc4b063fdc7f1edb8312de438722
Reviewed-on: https://webrtc-review.googlesource.com/c/107705
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25360}
2018-10-25 12:15:43 +00:00
23524ced41 Add HDR metadata struct
Bug: webrtc:8651
Change-Id: Ie7e263e945eedb47776e36e2e817991977e6ef6d
Reviewed-on: https://webrtc-review.googlesource.com/c/107709
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25359}
2018-10-25 12:04:37 +00:00
977b46a59b Export symbols needed by the Chromium component build (part 7).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I1081af5ecf7ba55a7415e09e45357b783cf300aa
Reviewed-on: https://webrtc-review.googlesource.com/c/107708
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25358}
2018-10-25 11:41:16 +00:00
3eb1c72bb6 Removes deprecated BitrateAllocation alias.
Bug: webrtc:9883
Change-Id: Ia14727a43c31241590889e48aded63dd8b30e181
Reviewed-on: https://webrtc-review.googlesource.com/c/107734
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25357}
2018-10-25 11:02:58 +00:00
2506839dba Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated.
Bug: webrtc:7510
Change-Id: Idfe645aa75cf6a0699caa94063f47c57c2ed5ee2
Reviewed-on: https://webrtc-review.googlesource.com/c/107728
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25356}
2018-10-25 10:59:57 +00:00
370bae466c APM: Adding more explicit handling of failures in the json config data
This CL creates a new API for the parser of APM json config that
that provides an explicit way for the user to know when there has
been an issue in the parsing of the json config data.

Bug: webrtc:9921
Change-Id: Idd8f40529f40ab6871efb5b356c0fd2cea21b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/107841
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25355}
2018-10-25 10:31:54 +00:00
487e694782 Use default value if field trial switch is set to an invalid number
Bug: webrtc:9851
Change-Id: I195e2e9b30905bd65f703098db9a1e7e44eac073
Reviewed-on: https://webrtc-review.googlesource.com/c/107620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25354}
2018-10-25 10:14:11 +00:00
273c851799 Remove obsolete android ndk copy from //third_party/android_tools/ndk
It lives at //third_party/android_ndk

This is from chromium repo
4d66cf2db5

Bug: None
Change-Id: Ia45355f1a7566a0b5268f7a3f33ca8e7dbe43451
Reviewed-on: https://webrtc-review.googlesource.com/c/107740
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25353}
2018-10-25 09:27:06 +00:00
7a95e0fcf4 APM: Add ability to turn on/off dumping of internal data
This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.

Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}
2018-10-25 09:03:53 +00:00
e2fd86a79c Move encoder metadata into EncoderInfo struct.
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
2018-10-25 08:51:53 +00:00
2d3a1fb950 Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
TBR: phoglund@webrtc.org
No-Try: True
Bug: None
Change-Id: I8c47031560c4e93c1cd3b28530ae64e06ee5399a
Reviewed-on: https://webrtc-review.googlesource.com/c/107717
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25350}
2018-10-25 08:19:53 +00:00
4191a81065 Revert "Move relay server code to a test-only target p2p_server_utils."
This reverts commit e284c521f76d810e9c68a238e4821e8f0f99a2cd.

Reason for revert: Breaks downstream project

Original change's description:
> Move relay server code to a test-only target p2p_server_utils.
> 
> Bug: webrtc:9798
> Change-Id: I5926cbb11922c7bd1adfa2099431dc461cc63f20
> Reviewed-on: https://webrtc-review.googlesource.com/c/107361
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25347}

TBR=mbonadei@webrtc.org,nisse@webrtc.org,perkj@webrtc.org

Change-Id: I8e66b556f0be4979e5ef223d93c97b4f993ab2f9
No-Try: true
Bug: webrtc:9798
Reviewed-on: https://webrtc-review.googlesource.com/c/107737
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25349}
2018-10-25 08:00:02 +00:00
8fb20cefc7 Roll chromium_revision f57bd4785e..d68fb50e14 (602511:602627)
Change log: f57bd4785e..d68fb50e14
Full diff: f57bd4785e..d68fb50e14

Changed dependencies
* src/base: 6cc6ee9644..d51406bdb7
* src/ios: c73894adeb..d091f1dc17
* src/testing: f1cc888569..ae87b1ea91
* src/third_party: 93e2726dec..7722c28f55
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/bdcfce54a8..ed6fe0f638
* src/tools: 07793bb3fe..5cd811feb9
DEPS diff: f57bd4785e..d68fb50e14/DEPS

No update to Clang.

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

Change-Id: Ia6bdbae5a59a0f6e0412faf7bf589354146c2639
Reviewed-on: https://webrtc-review.googlesource.com/c/107801
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25348}
2018-10-25 07:45:19 +00:00
e284c521f7 Move relay server code to a test-only target p2p_server_utils.
Bug: webrtc:9798
Change-Id: I5926cbb11922c7bd1adfa2099431dc461cc63f20
Reviewed-on: https://webrtc-review.googlesource.com/c/107361
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25347}
2018-10-25 06:58:32 +00:00
01c68b8001 Roll chromium_revision b9a687f112..f57bd4785e (602396:602511)
Change log: b9a687f112..f57bd4785e
Full diff: b9a687f112..f57bd4785e

Changed dependencies
* src/base: ea4fe4627a..6cc6ee9644
* src/build: be8c7ed1f9..5a371bcc0e
* src/ios: 2ef0f7fc00..c73894adeb
* src/testing: 4d3904898c..f1cc888569
* src/third_party: c7794df2c9..93e2726dec
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/00455e0148..bdcfce54a8
* src/third_party/depot_tools: 879d5e3796..2b71832f6d
* src/third_party/gtest-parallel: fe7f791f14..e472187d11
* src/tools: ae41d59fe5..07793bb3fe
DEPS diff: b9a687f112..f57bd4785e/DEPS

No update to Clang.

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

Change-Id: I6bffc7906570ec2b6dbe39b3fa5b0ad56f1deb39
Reviewed-on: https://webrtc-review.googlesource.com/c/107695
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@{#25346}
2018-10-24 23:19:09 +00:00
09a49fab28 Roll chromium_revision 869181c2dc..b9a687f112 (602275:602396)
Change log: 869181c2dc..b9a687f112
Full diff: 869181c2dc..b9a687f112

Changed dependencies
* src/base: fc75e9da03..ea4fe4627a
* src/build: 4b0fe3afe2..be8c7ed1f9
* src/ios: b480083c51..2ef0f7fc00
* src/testing: a3a1f924e0..4d3904898c
* src/third_party: 106ec94e49..c7794df2c9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/685b5de113..00455e0148
* src/tools: 2444eb8ba4..ae41d59fe5
DEPS diff: 869181c2dc..b9a687f112/DEPS

No update to Clang.

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

Change-Id: I278ba77d405cc77bd92b3578a582d5458a8f630b
Reviewed-on: https://webrtc-review.googlesource.com/c/107689
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25345}
2018-10-24 18:48:23 +00:00
4cb4786d78 Add expected default values to video configuration tests.
Bug: webrtc:9860
Change-Id: I772edf3c143eafaa97a37e596f3ea5a41f00b800
Reviewed-on: https://webrtc-review.googlesource.com/c/107676
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25344}
2018-10-24 16:45:38 +00:00
44a262a6aa Declares BitrateAllocator methods const.
Bug: webrtc:9883
Change-Id: I79e4f8d233281975c4073a381c7568469390b817
Reviewed-on: https://webrtc-review.googlesource.com/c/107725
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25343}
2018-10-24 15:26:36 +00:00
583d6d9d4f Add missing directory to api/DEPS and PRESUBMIT.py.
TBR: kwiberg@webrtc.org
Bug: webrtc:9887
Change-Id: Ib285005fc2a25549e72922bc38b05a170c6ef228
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/107707
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#25342}
2018-10-24 14:03:04 +00:00
62ae178357 Remove deprecated pipe field from VideoQualityTestFixtureInterface::Params
To be landed after 23th October

Bug: webrtc:9630
Change-Id: I8de460d093438c8b72bca44cdfce49b72cbcc2d0
Reviewed-on: https://webrtc-review.googlesource.com/c/104481
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25341}
2018-10-24 13:21:28 +00:00
825f83b99e Revert "Encode RTC event logs in new format."
This reverts commit ece3c228a2cbd1c1b05eee3a7f55dbb6f020acbc.

Reason: Breaks downstream project.

Bug: webrtc:8111
Change-Id: Ia264802b35a576d74b8a249ed742a8177e5cbe24
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107721
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25340}
2018-10-24 13:18:03 +00:00
e943d43926 Remove deprecated DefaultNetworkSimulationConfig
To be landed after 23th October

Bug: webrtc:9630
Change-Id: Ie322fe5428824b29ad51edaaa446121c5511b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/104600
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25339}
2018-10-24 12:57:31 +00:00
a418e67c53 Use checkdeps to ensure API headers don't include internal headers.
This CL updates the checkdeps configuration for the api/ folder in
order to explicitly avoid to #include non API headers from API headers.

In order to force a careful review of potential exceptions to this
rule, the CL also adds mbonadei@ and kwiberg@ as OWNERS of api/DEPS.

Bug: webrtc:9887
Change-Id: I0ada6f1020186b2782c7d060af36079c452ba1aa
Reviewed-on: https://webrtc-review.googlesource.com/c/106800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25338}
2018-10-24 12:55:01 +00:00
ec9b77bc42 Remove deprecated API: NetwrokSimulationInterface.
To be landed after 23th October

Bug: webrtc:9630
Change-Id: Ibf9c09d16e86789284491b16812ce57a3cad0624
Reviewed-on: https://webrtc-review.googlesource.com/c/104061
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25337}
2018-10-24 12:52:51 +00:00
257ed437f0 Add support for optional fields in FixedLengthDeltaEncoder
Optional fields are those which only occur sometimes. For example,
the sequence number field in an RTP packet always occurs, but
fields in optional RTP extensions only occur sometimes.

Bug: webrtc:8111
Change-Id: Iff2c35b73530c0a1db68e547b4caf34434aa4ace
Reviewed-on: https://webrtc-review.googlesource.com/c/103362
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25336}
2018-10-24 12:48:44 +00:00
c6ec4b1fa9 Fix w3c URL for RTCIceTransport
No-Try: true
Tbr: hta@webrtc.org
Bug: None
Change-Id: I51a6ad8c30e19cf999c5356e619770cfeee0068f
Reviewed-on: https://webrtc-review.googlesource.com/c/107638
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25335}
2018-10-24 12:36:09 +00:00
5e58bcbf29 Forward audio rtp frequency to Rtcp sender and use it for SR packets
Process video rtp frequency in the same way.

Bug: webrtc:6458
Change-Id: Ia22768e1242d686c2b3e2b911f3e5e492cf8b895
Reviewed-on: https://webrtc-review.googlesource.com/c/107651
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25334}
2018-10-24 12:27:09 +00:00
ece3c228a2 Encode RTC event logs in new format.
This CL adds the encoder and wires it up to the event log.
Parser and unit tests are uploaded in a separate CL.

Bug: webrtc:8111
Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
Reviewed-on: https://webrtc-review.googlesource.com/c/106708
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25333}
2018-10-24 12:21:43 +00:00
fb5c1eca30 AEC3: Included missing parsing of config parameter
Bug: webrtc:9912,chromium:898462
Change-Id: I8efb60367964d3880ba15ffd18349abd288d7307
Reviewed-on: https://webrtc-review.googlesource.com/c/107654
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25332}
2018-10-24 11:38:40 +00:00
8e6749e0dd Improve fileutils_override implementation internal API.
Use absl::optional instead of special constant to show, that we failed
to get OutputPath in fileutils_override

Bug: webrtc:9792
Change-Id: Ice19a9bf425e88a747dd9b07e82dbb5bdc59685b
Reviewed-on: https://webrtc-review.googlesource.com/c/107630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25331}
2018-10-24 10:24:14 +00:00
e068ad6262 Use a sufficiently large bitmask.
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.

Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
2018-10-24 09:27:18 +00:00
511fe0b2ca Roll chromium_revision 5e5003737d..869181c2dc (602066:602275)
Change log: 5e5003737d..869181c2dc
Full diff: 5e5003737d..869181c2dc

Changed dependencies
* src/base: 1fa2a3d59a..fc75e9da03
* src/build: 97454d191e..4b0fe3afe2
* src/ios: 25ccb1048b..b480083c51
* src/testing: fdf8fc0704..a3a1f924e0
* src/third_party: 49cf955ddc..106ec94e49
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/df56c1dae1..685b5de113
* src/third_party/depot_tools: 03d6d11896..879d5e3796
* src/tools: 6b4b60ca40..2444eb8ba4
DEPS diff: 5e5003737d..869181c2dc/DEPS

No update to Clang.

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

Change-Id: I2c3cbff203aa19db8ebb136405b7184bcfa643d2
Reviewed-on: https://webrtc-review.googlesource.com/c/107678
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25329}
2018-10-24 08:31:11 +00:00
d38a2b860b Increase the UDP receive buffer for video
Lost packets have been seen in high-bitrate applications and increasing
the UDP receive buffer reduced the problems.

Bug: b/115713113
Change-Id: I671f528afeaea525150fdc2013f2b245778e5d16
Reviewed-on: https://webrtc-review.googlesource.com/c/107580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25328}
2018-10-24 07:54:12 +00:00
f0c449e3ff APM: Correct includes required for the data dumping functionality
Bug: webrtc:5298
Change-Id: Ia8b8e6a308f1812216651efaf0e2249e9d0cbfd8
Reviewed-on: https://webrtc-review.googlesource.com/c/107631
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25327}
2018-10-24 07:38:28 +00:00
700b4a4e65 AEC3: Allow limiting dominant nearend to the non-initial phase
This CL allows control over the dominant nearend functionality so that
it is not active during the initial phase, when estimates are less
certain.

Bug: webrtc:9906,chromium:898273
Change-Id: I5f61dac806ec3b1ebc1a3ec72f0a16d07a67f14a
Reviewed-on: https://webrtc-review.googlesource.com/c/107632
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25326}
2018-10-24 07:15:49 +00:00