Currently the stats callback is registered too early.
For now we ignore media transport for these callbacks (it was ignored
already), and we will introduce changes to media transport in the
future.
Bug: webrtc:9719
Bug: chromium:906998
Bug: chromium:906533
Change-Id: I24c0265d46ec2eb35743de6cd96a11d8c41fefbe
Reviewed-on: https://webrtc-review.googlesource.com/c/114904
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26062}
These legacy endpoints were supported with Plan B since the SDP
parser would fill in default MID values if a=mid was absent that
happened to match the default offered MIDs.
With Unified Plan, these default MIDs changed so the autofilled
MIDs do not match any more.
This CL adds information to the SessionDescription struct to
indicate whether or not a=mid was present and modified
PeerConnection::SetRemoteDescription to copy MIDs from the local
description if the a=mid lines are not present.
Bug: webrtc:9540
Change-Id: Ibf923b4ad59edb0facd06ddbd01cc10c62fc48e6
Reviewed-on: https://webrtc-review.googlesource.com/c/114820
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26054}
This adds support for a new DesktopCapturer::Callback method
OnDisplayChanged that is sent at the start of a desktop capture
session and whenever the display geometry changes.
This cl adds the basic structure to call this api at the start
of the capture session. Currently Windows only.
A follow-up cl will add support to call this whenever the display
geometry changes.
Bug: webrtc:10122, chromium:915411
Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
Reviewed-on: https://webrtc-review.googlesource.com/c/114020
Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26053}
This will be used to calculate a lower bound for the round trip time in
a later CL.
Bug: webrtc:9718
Change-Id: I0a1d22045961fe6bd343d1d6ce9b36490b036bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/114680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26050}
This replaces the current usage of AudioProcessing::level_estimator()
in that test.
The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.
Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
CallTest, VideoQualityTest and VideoAnalyzer used test::TestVideoCapturer
as an interface for video sources. Change to use VideoSourceInterface instead,
since that's all they need.
This is a preparation for making test::VcmCapturer usable as a
VideoTrackSource, and replace use of cricket::VideoCapturer in example code.
Bug: webrtc:6353
Change-Id: I445f5f6f9b7342230b89f53a5722df9c9e92834f
Reviewed-on: https://webrtc-review.googlesource.com/c/114881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26047}
This makes it possible to test custom network controllers without
requiring update to test framework. Also updating BBR performance
test to use this feature.
Bug: webrtc:9510
Change-Id: I0446de0403fe9d1f6dc3710c1d114887a6c359c5
Reviewed-on: https://webrtc-review.googlesource.com/c/114640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26046}
If we're in ALR, the acked rate is going to be significantly lower than
the current estimate for the link capacity. If we need to back off in
this situation (usually caused by latency spikes), this CL makes us back
off relative to current estimate if. We then immediately send a new
probe just in case the network did actually change.
All of this is behind experiment flags for now.
Bug: webrtc:10144
Change-Id: I062a259c36417eea2211d44592ef7fc979aa22b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26045}
This CL moves IncomingVideoStream::NewFrameTask closer to where it's
used and simplifies it somewhat. This makes it easier to follow the code
when debugging etc.
Bug: webrtc:9883
Change-Id: I359e2a5f4f2341259fd7e66a55c7a4b8bd9313ba
Reviewed-on: https://webrtc-review.googlesource.com/c/114720
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26041}
For somewhat similar funtionality, GoogCcNetworkController can
be used via GoogCcNetworkControllerFactory.
Bug: webrtc:9586
Change-Id: I298050184513f50c1b9ef5c21b8c9b7a6ca46fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/114543
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26040}
Classes AsyncProxyServerSocket, AsyncSSLServerSocket, and
AsyncSSLServerSocket are used only by test and example code.
Moved to server_socket_adapters.{cc,h}, and to the
rtc_base_tests_utils build target.
In the process, also deleted a few ancient and unattributed TODO
comments.
Bug: webrtc:9798
Change-Id: I21279c92bd8f1354fab7eeaf1f9697fedfc760e1
Reviewed-on: https://webrtc-review.googlesource.com/c/107735
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26039}
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
The class has been renamed to EncoderSimulcastProxy.
Bug: webrtc:10069
Change-Id: Ief03cfb27145798ac46692d9e51371d2e119eeb0
Reviewed-on: https://webrtc-review.googlesource.com/c/114551
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26031}
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
Using GetAudio events from SSRCs without incoming packets doesn't make sense, and should be prevented.
Bug: b/116685514
Change-Id: I48e38bb780549c71cb5f68d370a6819634ad487d
Reviewed-on: https://webrtc-review.googlesource.com/c/114321
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26017}
The removed coded causes problems if the same RTCP packet is forwarded
to the congestion controller multiple times.
Bug: webrtc:10125
Change-Id: I659d8f8f3ce3c643710156fa81176ceeaedd714a
Reviewed-on: https://webrtc-review.googlesource.com/c/114165
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26016}
