Start using RobustThoughputEstimator in DelayBasedBwe test in preparation for making it default.
Experiments has not showed significant metric changes. However, simulations has showed that RobustThroughputEstimator better follow the actually receive rate better. Especially during bursts of sent packets. Code is also simpler.
Bug: webrtc:13402 chromium:1411666
Change-Id: I83cfa1fc15486982b18cc22fbd0752ff59c1c1b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40644}
ChannelSendFrameTransformerDelegate::SendFrame() currently only
supports sending frames in a single direction. With this change, we
allow sending received audio frames.
Bug: chromium:1464847
Change-Id: I8113a3278dfce7b2ba709afecc672bc9af9c4a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316600
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40643}
This is mainly to remove them from the list of sigslot blockers.
Bug: webrtc:11943
Change-Id: I7908b953d7b2e3e1f7fd6c4da52412f70f1666c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317901
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40641}
Also run iwyu on async_udp_socket.cc
Bug: webrtc:11943
Change-Id: I4ca0f468d27be08fa869fde791aec51cf0029047
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317940
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40640}
Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame
We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.
Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
Remove redundant ssrc_ set, instead construct send_delay_stats_ early
Remove extra lookup when packet is sent out, instead memorize pointer to needed object
minor style improvments using syntax new in c++17
Bug: None
Change-Id: I0f0e28f5a01de0380502d4bee64cdf76e44a59cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317760
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40630}
In fucntion EncoderStreamFactory::CreateSimulcastOrConferenceModeScreenshareStreams, the follow code allows TL for H264.
const bool temporal_layers_supported =
absl::EqualsIgnoreCase(codec_name_, kVp8CodecName) ||
absl::EqualsIgnoreCase(codec_name_, kH264CodecName);
However, the helper function IsTemporalLayersSupported does not allow TL for H264. The diff unifies the logic by using the helper function
Bug: webrtc:15442
Change-Id: I1497ccc1cd5d3715310e0485f9179bd8e6948f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317542
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40629}
This is to ensure that a bad NetworkState estimate can not decrease BWE
unless an delay BW overuse has been detected.
Bug: webrtc:10489
Change-Id: Ic3a516345999eeba814200c2e295a19b347b2eb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317800
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40628}
Following https://webrtc-review.googlesource.com/c/src/+/262666, use_rtx() was checked in PeerConnectionFactory::GetRtpReceiverCapabilities but was missed in GetRtpSenderCapabilities. Therefore clients that hardcode use_rtx = false end up in an inconsistent state where RTX is not fully disabled.
Bug: None
Change-Id: Ice5f29a77c59e9081f9dd72c13c819024a34a7dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40625}
Instead, use BlockingCall to match with how unregistration is done.
This is needed because the ThreadWrapper implementation in Chromium, overriding the Thread implementation in WebRTC, does not order sent (blocking) tasks along with posted tasks.
That makes the functional difference that Thread1 posting and sending
tasks to Thread2, can not assume that the tasks run in the order they
were posted and sent. I.e. in this case:
// Running on Thread1.
thread2->PostTask([](){ Foo(); });
thread2->BlockingCall([](){ Bar(); });
Thread2 may actually execute Bar() first, and then Foo().
Bug: chromium:1470992
Change-Id: I1f83f12ce39c09279c0f2b3bc71c3a33e2cb16c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317700
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40624}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
The revised version should work in more network configurations.
Submitted with no-try to unbreak the build.
No-try: true
Bug: b/297247924, webrtc:12598
Change-Id: I4b4bc586af1ec2393dc257b9cebf06fd71268131
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40614}
Remove unused includes, including a TODO that is now irrelevant
Add missing includes
Remove definitinon for constexpr class constants as not needed since c++17 to avoid adding include for RTPExtensionType
Bug: webrtc:10198
Change-Id: I5f0ed15c5a9020d8b2e58bdfa213bb38eb59a840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317443
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40611}
The wrapped encoders may sometimes shift the callback
threads, so SequenceChecker is not legits for this case.
Replaced with a Mutex.
Bug: b/296242528
Change-Id: I7b2e6e630563246d5214ff4f18c6855ba7869a92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40609}
When GenericFrameDescriptor or DependencyDescriptor RTP extensions are used, we may receive multiple consecutive StapA packets where only the first packet has is_first_packet_in_frame set. The previous code assumed that all StapA had is_first_packet_in_frame = true. Per discussion on the attached bug, removing the DCHECK is OK.
Bug: webrtc:15155
Change-Id: I6348740eac7d70bca2b7541721aaa7e2b5e5a970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316941
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40608}
This should replace the wrapping async DNS resolver used
for default resolution.
Bug: webrtc:12598
Change-Id: Ic65ecd17da7a5695d0003178aeb30824a707ec78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316928
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40607}
Expose new function MaybeCreateFrameDumpingEncoderWrapper that
wraps another passed encoder and dumps its encoded frames out
into a unique IVF file into the directory specified by the
"WebRTC-EncoderDataDumpDirectory" field trial. If the passed
encoder is nullptr, or the field trial is not setup, the function
just returns the passed encoder. The directory specified by the
field trial parameter should be delimited by ';'.
The new function is wired up in VideoStreamEncoder.
Bug: b/296242528
Change-Id: I6143adf899f78fcc03d4239a86c68dcbab483f1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40600}
since that transport is unset most of the time when rtcp-mux is used.
BUG=None
Change-Id: Ic1d732369c5544059112173af767488aed7ec8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316926
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40598}
Goal is to be able to get an improved overview of the distribution
of the total delay.
Bug: None
Change-Id: I0dced53eafd1fb09941590f3706480066c52419b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40597}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
