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}
The metrics of interest are the pacer rate, both normally and during a
queue drain, as well as padding rate.
Bug: None
Change-Id: I9c36219f63ce61b46f20d42678e0d97cb2a1873c
Reviewed-on: https://webrtc-review.googlesource.com/c/123195
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26855}
This CL introduces the possibility to poll the 2 peer connections
at constant intervals.
It also introduces a dummy AudioQualityAnalyzer that will have to
be implemented in a follow-up CL and it moves every type of the
test framework inside the webrtc::test namespace.
Bug: webrtc:10138
Change-Id: I40acf7894bd67ea5229baba2d2cf18cd8ef65e67
Reviewed-on: https://webrtc-review.googlesource.com/c/123441
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26854}
The Legacy ADM has remained to be available for testing. Now we are
ready to move on to using only the Java ADM.
Bug: webrtc:7452
Change-Id: Ic95b04b933e165f3c16b587a44384a2c965ef16c
Reviewed-on: https://webrtc-review.googlesource.com/c/123921
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26852}
For binary size reasons; they emit too many instructions at each call
site.
I decided to not point out replacements for them, since rtc::join has
only one caller, and rtc::split isn't the only option for splitting
strings. And both of them have APIs that can cause the creation of
needless temporary strings.
Bug: none
Change-Id: I06161012836a90b09249c2813d2ddc15f167d580
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/124480
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26846}
RIDs will now appear in the sent packets when they are supplied.
This is relevant for the Simulcast scenario which uses RIDs to
identify the different layers.
Bug: webrtc:10074
Change-Id: I2f281abc144f467e151a30ec13b8c375be4ac3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/124140
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26843}
Currently the IceTransportState goes to connected before the connection
is writable and also goes to checking before any candidate pairs are
being checked. This change fixes these cases.
Bug: webrtc:10352, chromium:933802
Change-Id: I64a67c7f76a94a6be9da413740ddc8762fe966ce
Reviewed-on: https://webrtc-review.googlesource.com/c/124102
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26842}
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}
There seems to be a race caused by the libevent wrapping TaskQueue
implementation when reposting a repeated task at destruction time. This
race results in the posted task being leaked according to asan.
Bug: webrtc:10278
Change-Id: Ida40b884547f3f789a804ca0ab3ce36982a4d68e
Reviewed-on: https://webrtc-review.googlesource.com/c/121424
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26839}
This CL introduces the possibility to save an RTCEventLogs from the
call in order to do further analysis and call debugging.
Bug: webrtc:10138
Change-Id: If95ef66ecf52218b34ce01a4bcf8ab7991b04e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/123881
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26838}
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}
There are two RTT values reported to GoogCC. They come from the same
source initially but one is calculated and smoothed in the video call stats.
However, there's not really any technical reasons why this value should
be received via the stats, this has just been maintained for legacy reasons.
Experiments shows no real difference between the modes, therefore the
stats-reported RTT is removed in this CL as a cleanup.
Bug: None
Change-Id: If1462d6c91570ffb883ecef2ba034f04a571c9b5
Reviewed-on: https://webrtc-review.googlesource.com/c/123883
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26833}
For e.g. when audio receiver is recreated during SetRtpExtensionsAndRecreateStream in webrtc_voice_engine.h,
the audio minimum delay can't go down.
Imagine we set base minimum playout delay when audio receiver stream is created, then its value will be cached, to be applied during recreation. Then SetRtpExtensionsAndRecreateStream is fired, and audio receiver stream is recreated with the cached value, but currently it in the constructor it is used to initialize both base minimum playout delay and minimum playout delay. Which leads to the bug that effective minimum playout delay can't go down anymore as if you set base minimum playout delay to the low value then effective delay use the biggest value which minimum playout delay.
This didn't come up during previous trials because of
https://webrtc-review.googlesource.com/c/src/+/122280
It was reseting minimum playout delay to 0 asynchronously, that is why you couldn't see this bug.
Bug: webrtc:10287
Change-Id: I924446bfcb33ac94f7e5bf987a1868acaf1b0346
Reviewed-on: https://webrtc-review.googlesource.com/c/124000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26832}
This removes the temporary non-const method that was kept in the code to
enable backwards compatibility while we fix downstream project dependencies.
Bug: webrtc:10251
Change-Id: Ie221af1d3b0f19112449d61e0f357a833f7a8b18
Reviewed-on: https://webrtc-review.googlesource.com/c/123561
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26824}
This informs the media transport that PeerConnection wants to use a data channel
and gives it a chance to set up before the data channel sends the first message.
Bug: webrtc:9719
Change-Id: I6ea905a74b29b8735e77ac68bc8606e7bca77f18
Reviewed-on: https://webrtc-review.googlesource.com/c/124020
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26823}
Also updating the tests so they pass. The previous target was not part
of the targets that runs on the try bots, so they started breaking
during a recent refactor. By adding it to the common target we'll avoid
regression like this in the future.
Bug: None
Change-Id: I55391d454bd52964875818be39cdaa853360ae81
Reviewed-on: https://webrtc-review.googlesource.com/c/123199
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26820}
In the Discussion part of
https://developer.apple.com/documentation/metal/mtltexture/1515679-replaceregion
it seems like we should sync the calls to replaceRegion (inside
setupTexturesForFrame) in RTCMTLRenderer and not just the command
buffer.
This is a speculative fix for the linked bug, but we don't have any
clear repro case. Have done basic testing in AppRTCMobile and don't
see any obvious regressions, so might be worth trying.
Bug: webrtc:10024
Change-Id: Id6848691129fba8845f38c3dfe0ba53b9e5a27ce
Reviewed-on: https://webrtc-review.googlesource.com/c/123766
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26819}
Replaces use of field trials in PacedSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.
BUG: webrtc:10335
Change-Id: Ie8870d93d51e996e762f2c2de7545bad261b6bb7
Reviewed-on: https://webrtc-review.googlesource.com/c/123521
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26818}
TransportSequenceNumberV2 is an experimental feature that should
not be part of the default offer. However, if we receive an offer
with this extension we should respond that we support it.
Bug: webrtc:10264
Change-Id: Id2424d421361e5d71f3a608cb8f74b63645c264a
Reviewed-on: https://webrtc-review.googlesource.com/c/123783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26817}
absl/memory/memory.h needs to be included for absl::WrapUnique on the
C++17 mode build, as absl/types/optional.h has a conditional include of
memory.h.
Tbr: terelius@webrtc.org
Bug: chromium:752720
Change-Id: I7bdfb200f29ac8ec11f0b102304b7788595e7eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/123360
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26815}
This CL moves the functionality in VideoSender into VideoStreamEncoder
and simplifies the code where possible, given what we know of the
encoder state and that we now run on the encoder queue.
The intent here is to make it easier to remove the next parts, the
encoder database and generic encoder wrapper.
Bug: webrtc:10164
Change-Id: I8c108ccbe5db97cd9fd1e84228134709af845ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/123540
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26813}
Create LossNotificationController, which produces LossNotification
RTCP feedback messages when video packets/frames are lost.
(LossNotification messages are sent when an RTP gap is detected,
as well as when frames are later received which are undecodable
because of the missing frames due to the previously dropped packets.)
Bug: webrtc:10336
Change-Id: I7b3a156ed14e5a727349acdd82dae6997462421b
Reviewed-on: https://webrtc-review.googlesource.com/c/123762
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26812}
As a unintended consequence of the changed iceConnectionState implementation we won't ever transition to the "failed" iceConnection state unless a connection has first been established. This CL fixes that and adds a integration test for this scenario.
Bug: chromium:933786
Change-Id: I45effd7411959ac0e5b16a13d7568756dbeff4d1
Reviewed-on: https://webrtc-review.googlesource.com/c/123785
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26811}
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
Pretty-Fast Fast Fourier Transform is a 3rd party FFT C library meant to
replace other FFT libraries in WebRTC (see https://crbug.com/webrtc/9577).
This CL adds a WebRTC wrapper meant to be used inside the Audio Processing
Module (APM). As a first step, it only supports aligned memory allocated
via PFFFT. Support for the C++ standard library containers will be done
afterwards since it requires careful investigation and benchmarking (because
PFFFT uses SIMD optimizations).
The wrapper pre-allocates a scratch buffer to avoid VLA.
Bug: webrtc:9577
Change-Id: Ied00c3d3b1df292024f608ccf0ed1917d6e92e56
Reviewed-on: https://webrtc-review.googlesource.com/c/122563
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26808}
