Sender isn't actually require to identify the stream, so specifying it
every time is useless. This CL removes sender from StatsKey object and
introduces StreamsInfo object which contains all required metadata about
streams that are seen by DVQA.
Bug: b/205824594
Change-Id: I5b6be3865a30fd5980ff6e7e50906abe70a632ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238562
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35399}
It will ensure that clock moved forward, so clock related metrics as
CPU usage will be also calculated correctly. We should dig into it deeper
to fix the root cause later.
Example failure:
https://ci.chromium.org/ui/p/webrtc/builders/try/win_asan/44610/overview
Bug: b/205824594
Change-Id: If1ebcf6a2b88ba0054479be292cca8f50506e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35379}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.
This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.
Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.
[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc
Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
DefaultVideoQualityAnalyzerFramesComparator::Stop() may not block until
all frames comparisons are processed in case when new comparison was
added after worker thread checked for available comparisons and Stop()
was invoked before worker thread checked the state_.
Bug: webrtc:13277, webrtc:13240
Change-Id: Ic16fdc01e43c04529cd83e5d9ef66d7573973cfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235205
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35212}
A preparation for splitting server sockets out into a separate
interface, see https://webrtc-review.googlesource.com/c/src/+/232607.
Transition plan:
1. Land this cl.
2. Update downstream code to use the new name.
3. Attempt landing
https://webrtc-review.googlesource.com/c/src/+/232607. May need
additional steps to not break downstream implementations of
PacketSocketFactory::CreateServerTcpSocket.
Bug: webrtc:13065
Change-Id: Ife448c705222f4c9f66a096e3dc7eb07e0f9c3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35155}
Use config from FakeEncoder in some tests.
Bug: none
Change-Id: I1d7e01f604f8aabb5d6815bb519ef2532d024d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233243
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35138}
Unlike ReadBits, ConsumeBits doesn't limit number of bits it may advance,
and thus should work when that number is close to the integer limit
Bug: chromium:1250730
Change-Id: Ia7847869ef9d3fc16450d572c9e2be6e1aa36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232332
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35042}
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.
Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
This is tested by a simple unit test and a new fuzzer that verify that all that can be parsed also can be written.
Bug: webrtc:12000
Change-Id: I461aedf97d3dec6e8916e72110fa097c3b31c27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34986}
The VP9 encoder may drop a frame internally which will not advance the
frame pattern. Consider the following scenario where only spatial layer
0 and temporal layer 0 is active:
1. Key frame encoded
2. Spatial layer 1 is activated
3. Delta T0 dropped
4. Delta T0 encoded
No S1T0 frame is encoded in (1) since it's not active. When
NextFrameConfig is called in (3) it will say that future frames may
reference T0 on both S0 and S1, but it's then dropped.
On step (4), the SVC controller essentially thinks it's encoding a new
picture and will happily reference the T0 on what it thinks is the first
delta frame. However, this is actually still the key frame and since
there was no S1T0 frame produced it will reference an invalid buffer.
To fix this, only say it's possible to reference a T0 frame after it has
been successfully encoded.
Bug: webrtc:11999, webrtc:13142, chromium:1178444
Change-Id: Iab3d2042ce0b3fa7d952b2831d1a36b1a6613a86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231695
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34982}
With the new SVC controller this will hopefully help uncover more subtle
bugs.
Bug: webrtc:11999
Change-Id: Iab76d38b3fb8dfbbeb269f4ba1e74f6f425501f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231694
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34981}
So that applications don't need to construct it from the exposed
network_thread.
The EmulatedNetworkManagerInterface::network_thread() accessor is currently
used as a way to get to emulation's SocketServer, and should be deleted
when applications of the emulation framework have migrated away from
that usage.
Bug: webrtc:13145
Change-Id: I3efa55d117cad8ac601c48a9d2d2aa62a121f9c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231649
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34964}
This is to allow tests with more supported codecs even if a layered
codec is used.
Bug: none
Change-Id: I35b866993e8c3dd077ac5c0b566e15efcf4b41c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231500
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34949}
It can happen that SFU will resend the frame which was before
considered as dropped during stream switching.
Bug: b/197740434
Change-Id: I95a67e6e637f6005a24df15875b50133a6e8eaaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230423
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34865}
Deletes the helper methods SdpSetObserver and SdpCreateObserver,
replaced with observer classes where used, in peer_scenario_client.cc.
Deletes the class webrtc_sdp_obs_impl::SdpSetObserversInterface, which
indirectly inherits rtc::RefCountInterface twice. Migrates this code
to use rtc::make_ref_counted, and migrates away from deprecated
versions of SetLocalDescription and SetRemoteDescription that use raw
pointers and SetSessionDescriptionObserver.
Bug: webrtc:12701, webrtc:11798
Change-Id: I18ea3fb51f533d7454a6dc75292b1827b1c80ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229981
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34843}
