Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
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}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.
Reason for revert: Reland by removing the conflict with the broken CL.
Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
>
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
>
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
>
> This causes a merge conflict. So need to revert this first.
>
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> >
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> >
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> >
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org
Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
This causes a merge conflict. So need to revert this first.
Original change's description:
> Move allocation and rtp conversion logic out of payload router.
>
> Makes it easier to write tests, and allows for moving rtp module
> ownership into the payload router in the future.
>
> The RtpPayloadParams class is split into declaration and definition and
> moved into separate files.
>
> Bug: webrtc:9517
> Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> Reviewed-on: https://webrtc-review.googlesource.com/88564
> Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23983}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/88821
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23991}
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.
The RtpPayloadParams class is split into declaration and definition and
moved into separate files.
Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
This ensures the event logs in CallTest will be used by default.
Bug: webrtc:9510
Change-Id: I9df82b5ef39e1b2cba2789f8c5c7a9fed3c4c2f6
Reviewed-on: https://webrtc-review.googlesource.com/88562
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23970}
This prepares for replacing single instance members with vectors in a
follow up CL.
Bug: webrtc:9510
Change-Id: Ie05436ec89a0af9ce9fe9cece9842a39227246ec
Reviewed-on: https://webrtc-review.googlesource.com/88180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23968}
This makes it possible to use them in VideoQualityTest and prepares for
allowing saving logs in other tests as well.
Bug: webrtc:9510
Change-Id: I3b1cc187d88e4f17745414433c2f96efd836a302
Reviewed-on: https://webrtc-review.googlesource.com/88561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23964}
This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend.
Bug: webrtc:9517
Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7
Reviewed-on: https://webrtc-review.googlesource.com/88241
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23936}
Removes checks that are not relevant to the particular tests. The checks
create dependencies on the CallTest base class. This prepares for
further refactoring in CallTest.
Bug: webrtc:9510
Change-Id: Ie6b0093a8fcb8a152ca58f421727c4b085b60a87
Reviewed-on: https://webrtc-review.googlesource.com/87845
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23930}
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.
Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:
- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h
The following things are moved to API:
- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)
These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.
This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.
Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
Only user was the replay.cc tool, when dumping frames to a file. It is
changed to instead inject a special decoder.
Bug: None
Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f
Reviewed-on: https://webrtc-review.googlesource.com/83121
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23675}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This experiment is now wired up inside of chrome using field trial and
this passthrough is now obsolete.
Bug: chromium:794608
Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08
Reviewed-on: https://webrtc-review.googlesource.com/82780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23625}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
std::mem_fun is deprecated in C++11, and removed in C++17. Using C++17
option for building libwebrtc causes build failure. This is found during
upgrading WebKit tree from C++14 to C++17.
This patch replaces std::for_each and std::mem_fun with range-based-for.
We also merge loops for streams_ into one.
Bug: webrtc:9277
Change-Id: I44a7e44ea21fc33ffa9a586ddfea570f97dfacb6
Reviewed-on: https://webrtc-review.googlesource.com/77280
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23285}
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".
It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.
Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
This prepares for allowing injection of a network controller.
Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
This simplifies the code and removes the need for a lot of bookkeeping
variables.
Bug: webrtc:9232
Change-Id: I0c9a4b0741ed5353caa22ba5acdcb166357441f2
Reviewed-on: https://webrtc-review.googlesource.com/74240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23149}
Replaces both BitrateConstraintsMask and
PeerConnectionInterface::BitrateParameters. The latter is kept
temporarily for backwards compatibility.
Bug: None
Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f
Reviewed-on: https://webrtc-review.googlesource.com/74020
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23148}
This CL extracts the part of FakeNetworkPipe responsible for simulating
network behavior into the SimulatedNetwork class, which implements the
new FakeNetworkInterface.
This prepares for an upcoming CL where the network simulation can
be injected in FakeNetworkPipe, allowing custom simulation models to be
used.
Bug: None
Change-Id: I9b5fa0dd9ff1fd8ccd5a7ce2d9ea3a5b11c5215e
Reviewed-on: https://webrtc-review.googlesource.com/64405
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23146}
Moving ownership of worker task queue in Call to
RtpTransportControllerSend. This CL also ensures that the task queue
is not destroyed until the process thread running
SendSideCongestionController is stopped.
The worker queue should be owned by RtpTransportControllerSend since
it is mainly used for rtp and transport related tasks such as bitrate
allocation and signaling network state.
Bug: webrtc:9232
Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9
Reviewed-on: https://webrtc-review.googlesource.com/63023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23119}
To make this work, also make the private method
GetTransmissionMaxBitrateMultiplier static.
Bug: None
Change-Id: I1777602b89d404ecbc9956acf9a9c7f852d69911
Reviewed-on: https://webrtc-review.googlesource.com/72440
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23023}
This CL changes the usages of milliseconds as a unit in FakeNetworkPipe
to microseconds. This matches the time unit of the PacketTime struct and
increases the precision of the simulation. The time resolution in
FakeNetworkPipe::Config is kept unchanged to keep the values more human
readable.
This CL prepares refactoring in upcoming CLs.
Bug: webrtc:9054
Change-Id: I103f7a0afa41381f676ea07fcc8c083532e61f1d
Reviewed-on: https://webrtc-review.googlesource.com/64140
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23018}
This CL moves the responsibility for demuxing from FakeNetworkPipe
to DirectTransport. This makes the interface for FakeNetworkPipe more
consistent. It exposes fewer different interfaces for different usages.
It also means that any time degradations applied to the packets due in
FakeNetworkPipe in tests will now be propagated to Call in a more
realistic manner. Previously the time was set to uninitialized which
meant that Call filled in values based on the system clock.
Bug: webrtc:9054
Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0
Reviewed-on: https://webrtc-review.googlesource.com/64260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23017}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
