Note that this value will override the minimum delay that is used for audio/video sync.
Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
Test execution was broken by specifying pool.
No need to do it once we specify bot ids in included json configs.
Bug: webrtc:10047
Change-Id: Ica5b891b796eec69573cc39d1d72617a68169499
Reviewed-on: https://webrtc-review.googlesource.com/c/112129
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25804}
The new wire format doesn't have much effect on compression unless
the log is encoded in reasonably large batches.
PeerConnection has two functions to start logging; one which takes
an output period (or batch size) in milliseconds and one which uses
a default period instead. This CL changes the default batch size to
5 seconds if the the new format is enabled as a field trial.
Bug: webrtc:8111
Change-Id: I638f6114325251b6a9acf4f863afe2688a3b0522
Reviewed-on: https://webrtc-review.googlesource.com/c/112130
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25803}
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
* RTCStatsIntegrationTest.GetStatsFromCaller
* RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
This CL prevents the conflict to happen by explicitly
closing the PeerConnection.
Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
Color space information will be transmitted as uint8_t. It's therefore
necessary to convert from uint8_t to the corresponding enums.
Bug: webrtc:8651
Change-Id: Ib7e7f9f6b4d7e0c291d283822180144944f3ea1e
Reviewed-on: https://webrtc-review.googlesource.com/c/111757
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25797}
- Enable flexible mode in loopback tools and quality tests
- Ensure duplicate references are not set by the sender in video header
- Reset first active spatial layer on keyframe in encoder
- Make vp9 encoder to not generate spatial references for first active
layer with external reference control in svc flexible mode
Bug: webrtc:10049
Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/112080
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25795}
before this CL it was only configured when pacer is used.
This CL sets it also when pacer is not used.
Move block for setting TransmissionOffset/AbsoluteTime extensions after pacer_ check
to stress in pacer case there are set(overwritten) in another function.
Bug: None
Change-Id: I06a6dd6ec689a25439a75b3baa71340535cd1ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/112126
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25794}
Propose resolution of Issue 10011 : (GCC) build fails desktop_capturer.cc:66:66: error: ‘strncmp’ was not declared in this scope
Bug: webrtc:10011
Change-Id: I4afdfd96f8bbc8e39380a365138ab79e237568e3
Reviewed-on: https://webrtc-review.googlesource.com/c/111885
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25790}
std::is_trivially_* is not available on certain old STL
implementations. Using absl implementation will allow
maximized compatibility.
Bug: webrtc:10054
Change-Id: I17ed0fff44328b3d7c51d14e8c4470f1df0e66ad
Reviewed-on: https://webrtc-review.googlesource.com/c/111728
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25788}
The fuzzers detected a possible overflow in the multiplication of sum and gainQ10.
Since gainQ10 cannot be larger than 2048000 (see WebRtcIsac_kQGain2Levels) and sum cannot be larger than 2^16, a int64 is large enough to hold the result.
Bug: chromium:904909
Change-Id: Icb12821d4006aaaaf70a5735d2abd2b96f7a2f0e
Reviewed-on: https://webrtc-review.googlesource.com/c/111921
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25787}
This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.
Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
He works on video analysis code every now and then, and EngProd
isn't much help when reviewing here.
Bug: None
Change-Id: I30b5f12584305d17d4c6a9682790fd0eda67d867
Reviewed-on: https://webrtc-review.googlesource.com/c/111881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25783}
WinUWP cannot use the win task queue as post/peek message event loop
is not available. A replacement version written using stdlib compatible
with WinUWP is added as an alternative.
Change-Id: Ie9d6e6f11f395d1815d8f04633772a0c597ed30a
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/108520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25781}
This reverts commit 1e87b4f32b73526f9caaae2a7bccfbd0cd84dcb9.
Reason for revert: Breaks internal project
Original change's description:
> Replace the IceConnectionState implementation.
>
> PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
> Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
>
> Bug: webrtc:6145
> Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
> Reviewed-on: https://webrtc-review.googlesource.com/c/108780
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25773}
TBR=kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,jonasolsson@webrtc.org
Change-Id: Icc4368d120a4167286fa6ba2e884a3650b453eff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6145
Reviewed-on: https://webrtc-review.googlesource.com/c/111925
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25775}
PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
Bug: webrtc:6145
Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/108780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25773}
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.
Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
Pass encoded frames to packetizer immediately if encoder is configured
to drop whole superframe.
Bug: webrtc:9950
Change-Id: Iedee9618bb146307efd5a86cb35bf14b5e64b341
Reviewed-on: https://webrtc-review.googlesource.com/c/109002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25771}
This prepares for providing an additional implementation of delay based
rate control. By moving the probe controller, less code will have to be
added in the upcoming CL.
Bug: webrtc:9718
Change-Id: I64eb2c8f5f7950b6e9d209f110dc0a757c710b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/111860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25770}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This is to let ANA config proto to fully control it.
Bug: b/119788974
Change-Id: Ib7842f784bdf879cb7d753c7077ce845f435a379
Reviewed-on: https://webrtc-review.googlesource.com/c/111741
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25764}
RSID is only useful if we store the RSID header extension.
Since we don't do that at the moment, there is no need to
store RSID in the stream configs.
Bug: webrtc:8111
Change-Id: I978f335d05984346f225c4781a8bfaa228f3f4c8
Reviewed-on: https://webrtc-review.googlesource.com/c/111759
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25763}
