The additional constructor unnecessarily increases the complexity
of the class and other downstream classes.
Bug: none
Change-Id: Ied797feb3c982a50b7b47e65018cfc90ca90bf6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40671}
since the spec and implementation took a different route
BUG=chromium:1354101
Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40665}
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
Trying to calculate std::abs(min_int) results in an int overflow.
Original author: asmok@.
Bug: None
Change-Id: I984e9ba4f48411a583a55cc3f9c66c9a1cc8dc92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318120
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40660}
The build target that CreateFromCGImage() belongs to, desktop_capture_obj
is not visible externally. A utility header is created to make it accessible.
Bug: chromium:1471931
Change-Id: Ie40f39114d277dc4b62fe2ce95a6b0c7b61a3943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318123
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40659}
Keep the logic managing whether audio RTP timestamps have the random
start offset added or not inside ChannelSend, so that the
ChannelSendFrameTransformerDelegate doesn't need to worry about it.
Crucially, this means that frames moved between senders by encoded
transforms clients will always use the correct offset for the channel
where we actually get sent.
Also rename TS variables throughout both classes to be explicit over
whether the offset has been added or not.
Bug: chromium:1464847
Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40655}
This time, hit the BUILD files too (where possible).
Bug: webrtc:11943
Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40654}
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: I38c309f74e8e1322602196354545b3a465866263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318040
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40653}
for the known standard semantics FID (used by rtx) and
FEC-FR (used byFlexFEC) they should match the expected two SSRCs.
For the nonstandard SIM group this should be limited by the maximum
number of simulcast layers supported.
BUG=chromium:1459124
Change-Id: I7cc2417a3ab207658ec80e8d7e9984c1ae631f53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315323
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40652}
Also change return type for username_fragment() to be const& and not
create a copy.
Bug: none
Change-Id: I8591af3da54fc8a9784e13cb000c4e02c0cd2f40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317980
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40651}
since their content typically is not processed further.
BUG=webrtc:142258
Change-Id: I5bcfb6c3a6f3a301acb497b83f8a4dbc3023c5db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317603
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@{#40649}
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}
