Commit graph

10 commits

Author SHA1 Message Date
Stefan Holmer
92cacec5c3 Fix mistake in range validation of WebRTC-BweBackOffFactor.
Bug: webrtc:8212
Change-Id: I89f236099736d2706b25ccc955789449c8e34853
Reviewed-on: https://webrtc-review.googlesource.com/7860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20215}
2017-10-10 09:16:07 +00:00
Ivo Creusen
46ca2879e1 Reland of https://chromium-review.googlesource.com/c/external/webrtc/+/616724 under field trial.
Bug: webrtc:8105
Change-Id: I8c68e0f270b3bd5d8da28b8334d4689064f607f6
Reviewed-on: https://webrtc-review.googlesource.com/4920
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20205}
2017-10-09 11:07:57 +00:00
Stefan Holmer
ea00e483d1 Add field trials to configure the backoff factor and the trendline window of the BWE.
These field trials can be set with a string similar to:
WebRTC-BweWindowSizeInPackets/Enabled-150/WebRTC-BweBackOffFactor/Enabled-0.95/

BweWindowSizeInPackets
Number of packets which the delay-based BWE window is based on. A larger value means lower delay-sensitivity.
Default in WebRTC: 20
Reasonable values for streaming: 50-150

BweBackOffFactor
How far the BWE will back off when the delay increases. A value closer to 1.0 means smaller back-off.
Range: > 0.0, < 1.0
Default in WebRTC: 0.85
Reasonable values for streaming: 0.85-0.95

Bug: webrtc:8212
Change-Id: I61f0883788b689847a43273b63cef663042f4d42
Reviewed-on: https://webrtc-review.googlesource.com/6764
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20172}
2017-10-06 07:10:04 +00:00
Elad Alon
1d87b0e40f Create RtcEventLogEncoderLegacy
We're moving to an RtcEventLog interface that accepts std::unique_ptr<EventLog> and stores the event for encoding when encoding becomes necessary, rather than before. This will be useful while we maintain the legacy (current) encoding alongside the new encoding on which we're working.

This CL introduces RtcEventLogEncoderLegacy, which takes provides the encoding currently done by RtcEventLogImpl. After this, we can modify RtcEventLogImpl to use a dynamically chosen encoding, allowing us to easily choose between the current encoding and the new one on which we're working.

BUG=webrtc:8111
TBR=stefan@webrtc.org

Change-Id: I3dde7e222a40a117549a094a59b04219467f490a
Reviewed-on: https://webrtc-review.googlesource.com/1364
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20116}
2017-10-03 13:51:59 +00:00
Danil Chapovalov
599df85233 Resolve cyclic dependency in remote bitrate estimator
Access SendTransportFeedback function through new interface to break rbe -> pacing -> rbe cycle
Depend on rtp_rtcp_format source set to break rbe -> rtp_rtcp -> rbe cycle.

Bug: webrtc:6828
Change-Id: Iae1c463a71871c0055485e2eca9b2235d770afec
Reviewed-on: https://webrtc-review.googlesource.com/1620
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19947}
2017-09-25 15:10:14 +00:00
alexnarest
b335e31bcb This is a rollback of https://chromium-review.googlesource.com/c/external/webrtc/+/616724
it degraded results of the ANA testing

BUG=webrtc:8105

Review-Url: https://codereview.webrtc.org/3011323002
Cr-Commit-Position: refs/heads/master@{#19902}
2017-09-19 19:00:32 +00:00
charujain
cb728ea83a Fix Gn Untracked headers in webrtc/modules/video_coding.
Fixed following headers in this CL
===================================

src/webrtc/modules/video_coding/sequence_number_util.h
src/webrtc/modules/video_coding/codecs/interface/common_constants.h
src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h

src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h

src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h

src/webrtc/modules/video_coding/test/test_util.h
src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h

Remaining:
===========
src/webrtc/modules/video_coding/include/video_codec_interface.h
src/webrtc/modules/video_coding/include/video_error_codes.h

BUG=webrtc:7620

Review-Url: https://codereview.webrtc.org/3012323002
Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 10:08:08 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00