As an alternative to attaching custom array of bytes.
Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.
Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.
The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.
Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
This way it can be forwarded to lower layers. This makes it easier to
add information without having to change signatures of intermediate
classes. This will be used in a later CL to use the link capacity in the
Opus decoder.
Bug: webrtc:9718
Change-Id: I4a4c9d104fedb0e4a0bb7f14d169475940edbf7e
Reviewed-on: https://webrtc-review.googlesource.com/c/111508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25738}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
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: Ia58a3b4f3becf9e620d3991da8451d81f32f8ad0
Reviewed-on: https://webrtc-review.googlesource.com/90406
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24118}
bugs.webrtc.org/8439 introduces application data that could e.g. contain
timestamps. We would like to take different actions for this data
depending on whether this is the first time a packet is being sent.
Bug: webrtc:8906
Change-Id: Ib370d76beec2960d961bf44391930faa4b193479
Reviewed-on: https://webrtc-review.googlesource.com/77643
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Petter Strandmark <strandmark@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23426}
Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.
Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
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}
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}
