It's used for driving the old jitter buffer, which is used only when
vcm::VideoReceiver is used via the legacy VideoCodingModule api.
Bug: webrtc:7408
Change-Id: I179d5b26e112d9f94615d2e1b410b51a657aa05b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127294
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27147}
The video decoder thread is the pilot user.
For now this is an Android-only feature, since that's the only
platform we can print stack traces on.
Bug: webrtc:9987
Change-Id: Ie638c619673b5f159d91a32683fd787baf46479a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126222
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27127}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: Iedb3be944828a1caba55bbbd4dc0b56c55bbb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127624
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27123}
A recent refactoring introduced a subtle difference in how encoded
frame timing is measured. See
https://webrtc-review.googlesource.com/c/src/+/124122/14/video/video_stream_encoder.cc#b1278
After that change, the encode was considered done after encoding was
done. The old behavior included the time needed to call stats and
the video sink, which might include video quality tests related tasks
and RTP packetization.
In order to preserve the old behavior I'm moving timestamping to after
packetization again.
Note that the timing frame info still has a separate timestamp that
does explicitly measure encode time. This is used by the experimental
new overuse detector, so the effect of this change will be transient
anyhow.
Bug: chromium:941457, webrtc:10164
Change-Id: Ia990a1ceaeaf2c45d5df2a32d4f017cdb08e3c55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127569
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27108}
This is a reland of 184f6d5d75c198cb7b70b8f9b75e0b5096c6e577.
Incorrect build dependencies in downstream tests have been fixed,
and an initialization bug in this CL has also been fixed.
Original change's description:
> Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video.
>
> This allows external users of this test fixture to specify a custom
> path, rather than just a custom file name.
>
> Bug: webrtc:10349
> Change-Id: I84e886c8bc28583017ce9ed7b9e7ee6a8e95730f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126227
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27033}
TBR: kwiberg@webrtc.org
Bug: webrtc:10349
Change-Id: I0ec9dd26cd96c3db8ac8482893a26e62a1b1eefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127181
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27102}
Specifying relative path like "~" somehow doesn't work in some cases.
To pass an absolute path some parameter rewriting is required.
Also, as the stream gets recreated several times at the beginning of the
call, append the time to the filename to make them unique.
Bug: none
Change-Id: I1c914f9081adb4ac5c34584d96e542e8a863547b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27101}
It's routed to the old jitter buffer, which is instantiated but
unused.
Bug: webrtc:7408
Change-Id: I8fe2d8a2bfa6c2119945cc294d17982af4f830cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127541
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27100}
This prepares from removing the overload in a followup CL.
Bug: webrtc:10365
Change-Id: I80db16e7d37944e3dc7d2799bbf45ef8f439a22c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126860
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27091}
This is not used in practice as there's functionality on
other levels that serves the same purpose.
Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
This allows external users of this test fixture to specify a custom
path, rather than just a custom file name.
Bug: webrtc:10349
Change-Id: I84e886c8bc28583017ce9ed7b9e7ee6a8e95730f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126227
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27033}
In Chrome we sometimes get frames with native handlers all the time.
To enable variable frame rate we need to trust at least empty updates.
Also, trust empty update rect when scaling.
Bug: webrtc:10310
Change-Id: I6087b66d5e6138290a7c143f85ba9bc427ae40a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126223
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27021}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
This prepares for making AudioSendStream use its own task queue. In the
future more of the functionality that depends on running on the task
queue is planned to be moved directly into RtpTransportControllerSend.
They should instead get it from the transport controller. This affects
the media transport tests which previously assumed that the transport
controller could be missing. However, this is not something that is used
in production, so this is an improvement of the tests as they will
behave more like production code.
Bug: webrtc:9883
Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27010}
This CL plumbs an additional signal from VideoSendStream down to
VideoStreamEncoder, namely the amount of headroom that's left between
the encoder max bitrate and the current bitrate allocation for the
media track.
This will be used in follow-up CLs to tune encoder rate adjustment
and some codec specific paramaters a bit differently, based on the
knowledge if we are network constrained or not.
Bug: webrtc:10155
Change-Id: Ic6ccc79be5c6845468bab65b4ca9918b56923fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125981
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27008}
If minQP is reached and encoder undershoot consistently, we consider the
quality good enough and throttle encode frame rate.
Bug: webrtc:10310
Change-Id: Ifd07280040dd67ef6e544efdd4619d47bff951e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125461
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27003}
If minQP is reached and encoder undershoot consistently, we consider the
quality good enough and throttle encode frame rate.
This CL also adds perf tests for high fps vp9 screenshare.
Bug: webrtc:10310
Change-Id: I49fc7d31f9f596a9ecb5f85fe9e0c7861d4915f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125761
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26997}
to clearly signal passed ownership.
Drop support for accepting nullptr clock to avoid copying the Configuration structure.
Update all calls in webrtc to the new factory function
Bug: None
Change-Id: Ic5a78da8e59ba3988a757a9d9634fa31499ce0db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125901
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26994}
The old one has been deprecated for a long time.
Bug: webrtc:6333, webrtc:6898, webrtc:7861
Change-Id: Ib9b798262817e80019afcacc5b41d18957a28101
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124827
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26993}
Ignore rtc_link_task_queue_impl flag,
instead use build_with_chromium for custom chromium implementation injection
This changes TaskQueue implementation used in webrtc fuzzers in chromium:
from own webrtc implementation to chromium's.
Bug: webrtc:10191
Change-Id: I63be28b680ae8ea8ee1dbf0c699263c392ce29d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125196
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26977}
This test is verified to better catch the performance issues that were
fixed in issue 10275.
Bug: webrtc:10275, webrtc:10070
Change-Id: I4654f013b0fa08015af8572269b9df979e5a641f
Reviewed-on: https://webrtc-review.googlesource.com/c/125300
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26944}
The current Key Frame request system doesn't take into account failed
decryptions and this can lead to WebRTC spamming new key frame requests when
the issue is actually in the decryptor layer. To prevent this if frame
decryption is required for the PeerConnection key frame requests will not be
sent at 200ms intervals but will wait until the stream is decryptable before
utilizing this logic.
Bug: webrtc:10330
Change-Id: I188a21dfd142dec6175d9def95f39a2bc517017c
Reviewed-on: https://webrtc-review.googlesource.com/c/123414
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26931}
This prepares for future cleanup of how RtpTransportControllerSend is
used.
Bug: webrtc:10365
Change-Id: Idefc7e60f83819627c83b397949c8434d93491b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26923}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
Simulcast screenshare appears broken due to unrelated changes. It
implicitly relied on SimulcastEncoderAdapter fallback, which happened before
if streams had same resolution. It's not the case anymore. Thus, this CL
adds checks for different frame-rate in simulcast streams.
FullStackTests are also updated to use actual parameters.
Bug: none
Change-Id: I2c1ddb1b39edb96464a0915dfcb9cb4e18844187
Reviewed-on: https://webrtc-review.googlesource.com/c/124494
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26869}
This CL takes a few parts of VCMEncodedFrameCallback and
VCMGenericEncoder and folds some aspect directly into
VideoStreamEncoder. Parts related to timing frames are extracted
into a new class FrameEncodeTimer that explicitly handles that.
Bug: webrtc:10164
Change-Id: I9b26f734473b659e4093c84c09fb0ed441290e40
Reviewed-on: https://webrtc-review.googlesource.com/c/124122
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26862}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/123920
Patch set 1 is identical to the previous CL, additional patch sets fix
the bug that was introduced and adds test coverage.
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: Ieaf23457d69af0d6356b70461112892b14760b19
Reviewed-on: https://webrtc-review.googlesource.com/c/124488
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26857}
This reverts commit 715c4765b1ac20017e6e3b8b925d02536c6610c3.
Reason for revert: Breaks WebRTC roll to Chromium.
https://chromium-review.googlesource.com/c/chromium/src/+/1484629
# Fatal error in: ../../third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc, line 796
# last system error: 0
# Check failed: diff_ms >= static_cast<int64_t>(0) (-307 vs. 0)
#
Original change's description:
> Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder
>
> Since this "data base" only holds a single encoder instance it just
> serves to confuse object ownership. Removing it and giving ownership
> of generic encoder instance to VideoStreamEncoder.
>
> This CL also removes VideoSender interface from video_coding_impl.h,
> which is mostly a leftover from
> https://webrtc-review.googlesource.com/c/src/+/123540
>
> Bug: webrtc:10164
> Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
> Reviewed-on: https://webrtc-review.googlesource.com/c/123920
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26835}
TBR=brandtr@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: I5432878c4c2e497cd848c4ce1b190e0307df03ca
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/124402
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26841}
* LossNotificationController is the class that decides when to issue
LossNotification RTCP messages.
* RtpRtcp handles the technicalities of producing RTCP messages.
Bug: webrtc:10336
Change-Id: I292536257a984ca85d21d9cfa38e7ff2569cbb39
Reviewed-on: https://webrtc-review.googlesource.com/c/124123
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26840}
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/123920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26835}
The new name fits better.
Bug: None
Change-Id: I1f201ff07915ed6c18efeefb7380e2b286742bb9
Reviewed-on: https://webrtc-review.googlesource.com/c/123800
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26814}
