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}
StringToNumber is directly used in parsing the SDP so it should be fuzzed.
Bug: webrtc:10395
Change-Id: I85b520fbefd34d3dba49950c5ff297b482c572b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127123
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27089}
The generic video depacketizer was missed in the initial fuzzing pass.
Bug: webrtc:10395
Change-Id: I166f27fc5897a2eafe38dad8e074834fefcc330e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127041
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27088}
This simple fuzzer is intended to detect potential issues in the field trial
parsing code. Since these can be set by the browser it is better to have some
fuzzing coverage around this area.
Bug: webrtc:10395
Change-Id: I1b8b859d2107a0bc99cb7520cf0ef96f3d110547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127121
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27087}
Introduce a handle for route created with network emulation layer,
that can be used to remove it in future properly.
Bug: webrtc:10138
Change-Id: I9fb847caeee24333bafb328727711af005b09224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127283
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27074}
Also fix minor issues in this class.
Bug: webrtc:10138
Change-Id: Icb3ec7f6296c34da260e701ec51d7b87ce62a4d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127281
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27073}
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}
The fuzzer times out on too long inputs.
This CL limits tests to 400 000 bytes, ~ 12 seconds of 8 kHz float audio.
Bug: chromium:940209
Change-Id: I86b772f9d8989a8b129d933d25ece3631a6a365f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126780
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27059}
This CL is preparation for extraction of public API for network
emulation layer.
Bug: webrtc:10138
Change-Id: Id59204ea20a103dafce4122c59e51a354836c374
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126624
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27050}
GN complains when this BUILD.gn file is pulled in and both is_ios and
rtc_include_tests are false.
Bug: None
Change-Id: Ic637063a9dd25feb53595eeedfa4f22061ba34f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126231
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27025}
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}
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}
Subclasses of FakeEncoder need to fill out the CodecSpecificInfo and
RTPFragmentationHeader, and they also write to the encoded data of the
EncodedImage. This used to be done by subclasses chaining onto the
parent's OnEncodedImage callback, but that's not so nice, since the
EncodedImage argument is passed as a const ref there.
This change introduces a protected method EncodeHook for this purpose.
FakeEncoder calls this prior to calling OnEncodedImage, with non-const
pointers.
In addition, change FakeEncoder to use EncodedImage::Allocate, rather
than explicit new and delete.
Bug: webrtc:9378
Change-Id: Ie8182d1d5224aa3b7f15905612f6dbcebef0a555
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125880
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26988}
In this CL:
- Updated Vp8TemporalLayers::OnEncodeDone to take a CodecSpecificInfo
instead of a CodecSpecificInfoVP8, so that both the VP8 specific and
generic information can be populated.
- Added structs to represent the GFD template structure.
- Added code to generate templates for video/screensharing.
Bug: webrtc:10342
Change-Id: I978f9d708597a6f86bbdc494e62acf7a7b400db3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123422
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26987}
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}
In QualityAnalyzingVideoEncoder all encoded images that belongs to
unrelated simulcast streams will be marked as to be discarded. So
to support simulcast streams QualityAnalyzingVideoDecoder have to return
black frames when all encoded images in received concatenated encoded
image are marked as to be discarded. Also QualityAnalyzingVideoDecoder
shouldn't pass such encoded image into VideoQualityAnalyzerInterface.
Bug: webrtc:10138
Change-Id: I0f793a7dc04b5d6b10949479bd074b2db86c5c6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125460
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26973}
This is a lightweight signalling, which tells that two
frames are the same if they are the same view of the same frame from the
same file, without comparing actual buffer contents and searching for
changed pixels.
Bug: webrtc:10310
Change-Id: I5c6ae571fdf4cab88466cde88fe7c7a78ae121cc
Reviewed-on: https://webrtc-review.googlesource.com/c/125099
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26951}
This avoids timeouts on the server.
Bug: chromium:935089
Change-Id: I8b46664a7cf4d5f14a76b5d034a67453e730eb9e
Reviewed-on: https://webrtc-review.googlesource.com/c/124484
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26940}
CodecSpecificInfo has a default constructor, so initializing by memset is not necessary and is in the way of adding non-trivial members.
Related chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/1495533
Bug: webrtc:10342
Change-Id: I36046f919f5fc34ea51de7288ff5c9cc0f2950b8
Reviewed-on: https://webrtc-review.googlesource.com/c/125093
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26924}
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}
To correctly send media from Bob to Alice, when Alice is calling Bob
we have to add transceivers for Bob's media to Alice first, because
it is forbidden to add new media sections into answer in Unified Plan,
so Alice's offer have to contain media sections for Bob's media tracks.
Bug: webrtc:10138
Change-Id: I8a5f19aa5ed6051a981472d5c79493362365f587
Reviewed-on: https://webrtc-review.googlesource.com/c/124492
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26892}
Introduce test case name for proper metrics reporting across different
parts of framework.
Bug: webrtc:10138
Change-Id: I7c501413ca2f2ee40314d988855dec0c28381c47
Reviewed-on: https://webrtc-review.googlesource.com/c/124740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26886}
In encoded image data injectors use the last bytes of the payload,
instead of the first.
Rewriting first bytes of the video frames payload caused problems
as somewhere first are used to check if the frame is key-frame
and also to parse QP values.
Bug: webrtc:10138
Change-Id: I59b7313ee54a33b31f842ec28ef8d831fe24eea5
Reviewed-on: https://webrtc-review.googlesource.com/c/124490
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26875}
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}
* 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}
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}
