Also drop unneeded dependencies and a #pragma.
Bug: webrtc:5876, webrtc:7660
Change-Id: I3a46eaf60591b00e43c0647a6eb758e5443de466
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128773
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27311}
It was used only in examples/peerconnection/server/peer_channel.cc,
for questionable utility.
Bug: webrtc:6663
Change-Id: I4047eb12f35615621dd0b34a694dead51c5fd20d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128869
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27279}
This CL introduces the TimeControllerInterface that provides timing
related functionality. Most notably it provides a TaskQueueFactory
and facilitates creation of ProcessThread.
Two implementations of the interface are provided, RealTimeController
and SimulatedTimeController.
This prepares for an upcoming CL using these in Scenario tests.
Bug: webrtc:10365
Change-Id: Id956a29628d7e2f53ecaedadd643a9f697329d2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127297
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27244}
This will be used in an upcoming CL.
Bug: webrtc:10365
Change-Id: Ic5f44fdb7579de994dd0896116573de6a46dfc00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27211}
This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor
TaskQueueForTest can be used instead in tests and keep the convenient constructor.
Also cleanup the TaskQueueForTest a bit:
move the class to webrtc namespace
add default constructor
disallow copy using language construct instead of macro
cleanup build dependencies
rename build target (to match move out of the rtc namespace)
Bug: webrtc:10284
Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27193}
Move PeerConnectionComponents when creating PeerConnectionDependencies
instead of passing them by pointer in test_peer.cc in PC e2e test
framework
Bug: webrtc:10138
Change-Id: I490f576c6af3eab42df04ba597945e66a87880e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128579
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27180}
Rename resolution_of_encoded_image into resolution_of_rendered_frame in
DefaultVideoQualityAnalyzer to make it consistent with the way, how it
is calculated.
Bug: webrtc:10138
Change-Id: Ibf89f08ac0646b57b4a6b8316cec1ed73bad02a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128576
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27179}
Use deque instead of list in DefaultVideoQualityAnalyzer for frame ids
in the single video stream.
Bug: webrtc:10138
Change-Id: Ie4f004b6f2aa5facf216551a12bdafcf3fcddfee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128574
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27178}
Reduce resolution of smoke test in PC E2E test framework to reduce load
on bots, cause this test isn't part of performance test binary.
Bug: webrtc:10138
Change-Id: I2c3758583c03e75be17bfef799a31f63357834c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128380
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27157}
This change integrates fuzzing support for RtpDumps in WebRTC. This allows
LibFuzzer to directly fuzz the RTP code path from packet arrival all the way
to actual decoding and rendering. It does this by replaying each RTP packet
in the RTPDump which can be mutated directly by the fuzzer.
For fuzzing support the RtpFileReader needs to support reading from a
buffer instead of an file. The test class requires FILE* for all its
parsing operations and is deeply coupled this way. I chose to solve this
problem at an OS level by using the tmpfile() option and copying the buffer
to the tmpfile(). fmemopen() is no available on most platforms so couldn't
be used as a generic solution. The additional copy isn't ideal but won't
be a bottleneck for the fuzzing.
In the future I plan for the fuzzers to read from a configuration file. But
given the current packaging strategy for fuzzers in WebRTC this isn't easy.
Bug: webrtc:9860
Change-Id: I2560120e82663f9e9fb5b9640e6a6d16f9c1a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126682
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27151}
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}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
This change simply calls through all code paths in the SSLCertificate interface
after passing in an untrusted PEM string. Corpus will follow in another CL.
Bug: webrtc:10395
Change-Id: I001642fa89a84ce01505780f5e76f01a0e46a785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127640
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27118}
Adding an example of a request to send simulcast (from the PC).
Adding an example of a request to receive simulcast (from the SFU).
Bug: webrtc:10409
Change-Id: I13b689621e2f89f8e00b7ee8bc542157ccebb873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127621
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27116}
rtp_header_parser currently has 0% fuzzing coverage. To improve this I have
added a basic fuzzer which fuzzes all of the available paths.
Bug: webrtc:10395
Change-Id: I30324b2bfa7629b0110527258b33b7e048e89fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27115}
This change adds a basic fuzzer to exercise parsing of SCTP messages.
Bug: webrtc:10395
Change-Id: I1fd7a8560add3463c1978ebcad30082ae31f2073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27113}
This function is called on each incoming RTCP payload.
Bug: webrtc:10395
Change-Id: I164746fe45912cc503565e77046b5d884e0204e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127122
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27110}
* Removing unused return values.
* Using TaskQueueForTest to do blocking calls.
* Improving naming.
This prepares for future work to run scenario tests in simulated time.
Bug: webrtc:10365
Change-Id: I2c100e9c20f4b85e85d7b455ea01944f6a14e08f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127561
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27105}
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}
Adds an implementation of the CoDel active queue management algorithm
to the network simulation. It is loosely based on CoDel pseudocode
from ACMQueue: https://queue.acm.org/appendices/codel.html
Bug: webrtc:9510
Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6
Reviewed-on: https://webrtc-review.googlesource.com/c/123581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26834}
The FMT 15 is not specific only to REMB or loss notification messages.
Rather, it is the Application Layer FB (AFB) of Psfb (Payload Specific
Feedback Messages).
See https://tools.ietf.org/html/rfc4585#section-6.3TBR=terelius@webrtc.org
Bug: webrtc:10336
Change-Id: I8cd27ef9ee044bf7b7e7c1bd1a53c1dae2d95006
Reviewed-on: https://webrtc-review.googlesource.com/c/123886
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26827}
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}
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.
Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
To support analyze of spatial layers we will continue sending them
into the network on encoder side, but will mark which should be then
discarded and which should be processed. On decoder side we will drop
layers, if they should be discarded and decode only parts, that
should be processed.
Bug: webrtc:10138
Change-Id: Ic8b8fe7787674c0ec49b879fcc29e54e8e3d787f
Reviewed-on: https://webrtc-review.googlesource.com/c/123185
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26784}
The values are available as part of the RTPVideoHeader member.
Bug: None
Change-Id: I832fffc449929badec3796d7096c9cdc0d43d344
Reviewed-on: https://webrtc-review.googlesource.com/c/123234
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26773}
The timestamps doesn't always match properly, currently causing
flakiness and crashes. Pending a better solution we'll assume that
no frames are lost.
Bug: webrtc:9510
Change-Id: I1b0a5025ac9a45c71b611bcddbbad7a8cf385e01
Reviewed-on: https://webrtc-review.googlesource.com/c/123483
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26772}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
This reverts commit 4f36b7a478.
Reason for revert: Failing tests fixed.
Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}
TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
Since WebRTC stats are collected on the signaling thread, this CL moves
the wait from the signaling thread to the main thread.
Bug: webrtc:10138
Change-Id: I0e554fe82e3a4afe66b45e53032b06d533f54a39
Reviewed-on: https://webrtc-review.googlesource.com/c/123228
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26746}
This reverts commit 389b1672a3.
Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}
TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.
For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.
Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
Start sending media from the peer when it's ICE connection state is
connected.
Bug: webrtc:10138
Change-Id: I9f5a1cd917317a3ebadd7c156563035b0bbecf2a
Reviewed-on: https://webrtc-review.googlesource.com/c/121956
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26698}
Removing simulcast stream support as it was broken.
Bug: webrtc:9510
Change-Id: I42ba285bbea81e6ffd5b1d1a1aec4e5eb0990b1e
Reviewed-on: https://webrtc-review.googlesource.com/c/123040
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26684}
There was a name collision with downstream test frameworks.
Bug: webrtc:9510
Change-Id: I7e37a8a54701ef4a47c687aec51f37523759f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123044
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26683}
This implementation won't support spatial layers and simulcast. It will
be added in next CLs.
Bug: webrtc:10138
Change-Id: I08baef36fb15b8d2d2fa222c761d40508de7ff61
Reviewed-on: https://webrtc-review.googlesource.com/c/121944
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26676}
In order to correctly close audio dump files, TestPeers have to be
destroyed after the call is finished.
Bug: webrtc:10138
Change-Id: I948e4e1844dfbffd1eef7926a4dd4d7631dbe632
Reviewed-on: https://webrtc-review.googlesource.com/c/122301
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26661}
The lock is unnecessary and potentially unsafe:
1) All gain_control accesses in AudioProcessingImpl happen - and are intended to happen - while holding the crit_capture_ lock, and all external API calls take the same lock once inside GainControlImpl.
2) If ProcessCaptureStreamLocked (locked by crit_capture) calls a gain_control function that takes crit_render, the mandated locking order (render before capture) is violated and we might get a deadlock with the render thread.
Bug: b/123456404
Change-Id: Id7a888827e347e5e1d50e2f87d90e8b68f52b7b8
Reviewed-on: https://webrtc-review.googlesource.com/c/122087
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26637}
This reverts commit 5054f54457.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 2
>
> Implement test utility for extracting changed part of video frames.
>
> Bug: webrtc:10152
> Change-Id: Iead052d2a18384aaa828cd7821be961b8614568e
> Reviewed-on: https://webrtc-review.googlesource.com/c/120407
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26496}
TBR=ilnik@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I80cae8a7d352b4ee67b42f5388fd8c1883ab2e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/122091
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26632}
Currently there's an implicit requirement that users of
SimulatedNetwork should call it repeatedly, even if the return value
of NextDeliveryTimeUs is unset.
With this change, it will indicate that there might be a delivery in
5 ms at any time there are packets in queue. Which results in unchanged
behavior compared to current usage but allows new users to expect
robust behavior.
Bug: webrtc:9510
Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069
Reviewed-on: https://webrtc-review.googlesource.com/c/120402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26617}
with the value that actually ends up being assigned here. There is no change in actual behavior.
Bug: None
Change-Id: I268c50a920a5d7e98909a9ec760fc80ca0718417
Reviewed-on: https://webrtc-review.googlesource.com/c/121540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26565}
This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.
It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump
It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.
Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
It doesn't do anything any more, so it should be removed.
Bug: webrtc:9586
Change-Id: I0b320b6ce4f480ff8cb59451db29bcc77b882b5f
Reviewed-on: https://webrtc-review.googlesource.com/c/120813
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26507}
Implement test utility for extracting changed part of video frames.
Bug: webrtc:10152
Change-Id: Iead052d2a18384aaa828cd7821be961b8614568e
Reviewed-on: https://webrtc-review.googlesource.com/c/120407
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26496}
Create audio stream instead of data channel to check compatibility of
network layer with PeerConnection. Replacement is done because there is
a data race inside data channel sctp transport. This CL will fix
bot behavior. Further data race investigation will be done in this
bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=10268
Bug: webrtc:10268, webrtc:10138
Change-Id: I4f7a1116c65dbf4a3508b7d81d654ccd320795f0
Reviewed-on: https://webrtc-review.googlesource.com/c/120807
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26495}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:10270
Change-Id: I316f9788adac954c52b0f9230881b872c54a7ac9
Reviewed-on: https://webrtc-review.googlesource.com/c/120348
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26482}
This CL contains network emulation layer and is a first part of landing
CL https://webrtc-review.googlesource.com/c/src/+/116663
Bug: webrtc:10138
Change-Id: If664b21e9df847aef8144d622d08fc7e9f6608da
Reviewed-on: https://webrtc-review.googlesource.com/c/120406
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26470}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
TestPeer represent single participant in the call and will own most
required for call objects.
TestPeer::CreateTestPeer is responsible for full setup of TestPeer and
allow to correctly inject media analyzers into call.
Bug: webrtc:10138
Change-Id: Ide7062004b0dc113b9c05181d8144797a3cc27a8
Reviewed-on: https://webrtc-review.googlesource.com/c/119941
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26464}
This CL applies clang-tidy's modernize-use-override [1] to the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:10252
Change-Id: I2bb8bd90fa8adb90aa33861fe7c788132a819a20
Reviewed-on: https://webrtc-review.googlesource.com/c/120412
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26461}
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)
Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
Add static factory method from FrameGenerator for FrameGeneratorCapturer
to be able to intercept generated frames in PC e2e test framework to
dump input video stream into file, if it was generated.
Bug: webrtc:10138
Change-Id: Iabecfaaef804111e0b19756cd676c1749757d9c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119947
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26424}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.
Also fixing some minor test suite bug found during investigation.
Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
VideoQualityAnalyzerInjectionHelper will be used to provide all required
entities to inject video quality analyzer into peer connection pipeline.
Bug: webrtc:10138
Change-Id: Iea7cf453311d809619839d5cf94b78a020ce9167
Reviewed-on: https://webrtc-review.googlesource.com/c/119642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26395}
This encoder will be used to inject VideoQualityAnalyzerInterface into
VideoEncoder, so it will be able to measure its metrics and also trace
frames from capturing on one peer side to rendering on another peer side.
The decoder will be used for the same purpose but in VideoDecoder pert.
Bug: webrtc:10138
Change-Id: Idf719753e3c0b3b1369ff206365bf0558705eb98
Reviewed-on: https://webrtc-review.googlesource.com/c/117363
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@{#26381}
The FrameGeneratorCapturer instances continue to live after
RunBaseTest() returns, and have their own internal task queues. This
means any class that listens for frames may be called after return
from RunBaseTest(), at which point they may be destroyed.
This CL makes sure we remove any capturer before returning.
A specific example of this problem is
VideoSendStreamTest.SuspendBelowMinBitrate
Bug: None
Change-Id: I857566301acce3e32c0888c7a1d2ee3470e6eb28
Reviewed-on: https://webrtc-review.googlesource.com/c/116684
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26375}
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.
Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.
It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.
Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
VideoFrameWriter is designed to accept webrtc::VideoFrame as input and
write it with Y4mFrameWriterImpl to the output file, transforming
webrtc::VideoFrame to the uint8_t* frame_buffer. VideoFrameWriter will
be used to write webrtc::VideoFrames during dumping input and output
video in peer connection level test framework and will be injected
in webrtc::test::FrameGenerator and rtc::VideoSinkInterface<VideoFrame>.
Bug: webrtc:10138
Change-Id: Iadec7d3ad66f226836acbebe070cf88ceb242f62
Reviewed-on: https://webrtc-review.googlesource.com/c/117200
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26305}
This analyzer will be used in implementatino of peer connection level
test framework before main analyzer will be implemented.
Bug: webrtc:10138
Change-Id: Ibb7c5cd94b0f07c6fc5a2415f04b0f0ae7ae75e2
Reviewed-on: https://webrtc-review.googlesource.com/c/117221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26301}
In this reland, I disabled high bitrate webrtc perf test on Android32.
This is a reland of 15df2774f4
Original change's description:
> This CL adds a fake codec factory in WebRTC that can be used in tests to
> produce target bitrate output.
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org
Bug: chromium:879723
Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9
Reviewed-on: https://webrtc-review.googlesource.com/c/117980
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26285}
This makes it possible to save log outputs from scenario tests to
either files or memory.
Bug: webrtc:9510
Change-Id: I883bd8240ab712d31d54118adf979041bd83481a
Reviewed-on: https://webrtc-review.googlesource.com/c/116321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26284}
Use size() accessor function. Also replace most nearby uses of _buffer
with data().
Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
It will be used to dump generated audio from TestAudioDeviceModule into
user defuned file in peer connection level test framework.
Bug: webrtc:10138
Change-Id: I6e3db36aaf1303ab148e8812937c4f9cd1b49315
Reviewed-on: https://webrtc-review.googlesource.com/c/117220
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26267}
This change adds a new way for test code to serialize the important information
from a VideoReceiveStream::Config so that it can be stored as configuration data
for WebRTC fuzzers. This code isn't included in the object itself as it is only
going to be used to generate new configurations for the fuzzer each time a new
error_correction or video format is added to WebRTC.
Bug: webrtc:10117
Change-Id: I9b6fb8e0345890ab16f6d319d91e4e316d1f2888
Reviewed-on: https://webrtc-review.googlesource.com/c/116920
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26255}
This change moves the configuration parser that converts a JSON representation
of the VideoStreamReceiver::Config structure into a native object into the test
directory so that it can be shared with the new corpus_generator utility that is
being built. This rtc_source_set will have an additional utility function added
in a subsequent CL that will allow the generation of a VideoStreamSender::Config
from a given VideoStreamReceiver::Config and visa versa.
Bug: webrtc:10117
Change-Id: I3035826f799f8d1fcdeaa76997391f030c855a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/116880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26252}
EncodedImageIdInjector is responsible for injection of frame id into
encoded image before it will be sent to the transport layer. It will
help to track video frame from capturing on 1st peer side to rendering
on 2nd peer side and will make it possible to calculate video quality
stats between these frames.
This CL also introduces two different implementations for injector:
1. DefaultEncodedImageIdInjector will prepend all encoded images with
extra data and then will restore them on another side. This injector
can work even if peers are running on different devices.
2. SingleProcessEncodedImageIdInjector can work only when all peers
are running in the same process, but won't use any extra data
to propagate frame id between peers, so it won't affect any
transport level metrics and bitrate estimator.
This CL is first part of new video quality analyzer for end-2-end
peer connection level test framework.
Bug: webrtc:10138
Change-Id: I77defc8e8c95cb244a695a9732980a47bd7a2e9b
Reviewed-on: https://webrtc-review.googlesource.com/c/116682
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26251}
I am planning to do a lot of work on adding additional fuzzing to WebRTC in Q1
of 2019. Given the limited number of available owners in this directory, and it
being non-prod I think this makes sense.
Bug: webrtc:10117
Change-Id: If2ad52fbce120c4d2bce51f4bfed99d83d78e6a5
Reviewed-on: https://webrtc-review.googlesource.com/c/117043
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26250}
This Config configuration will eventually replace the AudioProcessing::noise_suppression() interface.
This also introduces a proxy NoiseSuppression, returned by AudioProcessing::noise_suppression.
Without this proxy, ApplyConfig could overwrite NS settings for clients who currently use noise_suppression(). For example, the following code will not preserve the noise suppression level:
apm->noise_suppression()->set_level(NoiseSuppression::kHigh);
auto cfg = apm->GetConfig();
apm->ApplyConfig(cfg);
The NoiseSuppression instance returned by noise_suppression() has no way to update the config inside APM, so GetConfig() will return an out-of-date config which is then re-applied. This CL adds a proxy that makes this update, by forwarding Enable() and set_level() calls to ApplyConfig().
Drive-by change: AudioProcessing::Config substructs are reordered to mirror the capture processing pipeline.
Tested: Ran ToT and this CL builds of audioproc_f and verified identical settings/aecdumps.
Bug: webrtc:9947
Change-Id: I823eade894be115c254d656562564108b2b63b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/116521
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26248}
The change in
https://webrtc-review.googlesource.com/c/116683 made the fuzzer crash
at startup.
Bug: chromium:921050, webrtc:10185
Change-Id: Ie3eb26e12b4ae9b29c1c424af0d3eb287b5f1a73
Reviewed-on: https://webrtc-review.googlesource.com/c/117261
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26238}
This reverts commit 15df2774f4.
Reason for revert: It's causing the Android perf bots to fail. E.g.: https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20%28L%20Nexus4%29/6666
Original change's description:
> Add a high bitrate full stack test with fake codec
>
> This CL adds a fake codec factory in WebRTC that can be used in tests to
> produce target bitrate output.
>
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
>
> Bug: chromium:879723
> Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
> Reviewed-on: https://webrtc-review.googlesource.com/c/97185
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26182}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org
Change-Id: I33cd01ce345d81d66543f9be99750fa100760b5c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:879723
Reviewed-on: https://webrtc-review.googlesource.com/c/116785
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26192}
This CL adds a fake codec factory in WebRTC that can be used in tests to
produce target bitrate output.
We also add a high bitrate test that makes use of fake codec. This test assumes
ideal network conditions with target bandwidth being available and exercises
WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
Bug: chromium:879723
Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
Reviewed-on: https://webrtc-review.googlesource.com/c/97185
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26182}
Also introduce interface for video quality analyze and mock interface,
that then will be extended for audio quality analyze.
Bug: webrtc:10138
Change-Id: I0e3957fb2af1b12e796f154765580ddf562c7814
Reviewed-on: https://webrtc-review.googlesource.com/c/116500
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26157}
It is a step in the big refactoring to introduce new network emulation
layer for peer connection level e2e test, which will be based on system
sockets level injection.
Bug: webrtc:10138
Change-Id: Ie3854d22aa3eec289617bc432026ea670646556a
Reviewed-on: https://webrtc-review.googlesource.com/c/115943
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26137}
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.
Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
To be able to reuse VideoBroadcaster, that class needs to be
officially threadsafe. It already had the needed locks, but thread
checkers have to be deleted to allow calls to AddOrUpdateSink on
multiple threads (worker thread + encoder thread).
Bug: webrtc:6353, webrtc:10147
Change-Id: I16128ac205c566f09402b6f22587a340d9a983c1
Reviewed-on: https://webrtc-review.googlesource.com/c/115201
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26073}
The return value is not used. This change prepares for future
refactoring by removing the requirement that TryDeliverPacket must be
synchronous. Also renaming to DeliverPacket as we no longer need to
indicate the meaning of the return value.
Bug: webrtc:9510
Change-Id: I78536434b198fa7bf4df88b10d6add23684767f1
Reviewed-on: https://webrtc-review.googlesource.com/c/115181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26066}
CallTest, VideoQualityTest and VideoAnalyzer used test::TestVideoCapturer
as an interface for video sources. Change to use VideoSourceInterface instead,
since that's all they need.
This is a preparation for making test::VcmCapturer usable as a
VideoTrackSource, and replace use of cricket::VideoCapturer in example code.
Bug: webrtc:6353
Change-Id: I445f5f6f9b7342230b89f53a5722df9c9e92834f
Reviewed-on: https://webrtc-review.googlesource.com/c/114881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26047}
This makes it possible to test custom network controllers without
requiring update to test framework. Also updating BBR performance
test to use this feature.
Bug: webrtc:9510
Change-Id: I0446de0403fe9d1f6dc3710c1d114887a6c359c5
Reviewed-on: https://webrtc-review.googlesource.com/c/114640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26046}
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
This will print out the major events during a NetEq simulation.
Bug: b/116685514
Change-Id: Iab172e9a9115695b42c67628d5523c727359bb89
Reviewed-on: https://webrtc-review.googlesource.com/c/114320
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26019}
Configuring video decoding and rtp depacketization through json was introduced
in a prior change. This change introduces some basic configurations that will
be used in the initial round of fuzzers that are being added.
TBR=henrik.lundin@webrtc.org
Bug: webrtc:9599
Change-Id: I58aba6a6f24f8374126547deeef0ff4d1708327b
Reviewed-on: https://webrtc-review.googlesource.com/c/113834
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26005}
rtc::s_url_decode internally calls transform on rtc::url_decode which operates
on raw char buffers. This is used in some core parts of ice server parsing so
it makes sense to add at least a basic fuzzer here. Corpus generation will be
tailored in a future CL.
Bug: webrtc:10117
Change-Id: If1685601c746c4a9f88c2a8d396eeb3f1b1688d4
Reviewed-on: https://webrtc-review.googlesource.com/c/113835
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25980}
Some tests had to be updated due to this change.
Bug: webrtc:9510
Change-Id: I79c4c0166d8ba5e8190a607d5d35b67dc30a3c14
Reviewed-on: https://webrtc-review.googlesource.com/c/113522
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25952}
This change converts all tests but CodecInternalCng and
DecodingErrorDuringInternalCng, which depend on the obsolete Decode
method.
Bug: webrtc:10080
Change-Id: I34b068b3aa7139ed24bd63b417a5adcfc1de7922
Reviewed-on: https://webrtc-review.googlesource.com/c/113506
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25946}
This reverts commit cdc5eb0de1.
Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().
Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
>
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
>
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}
TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.
Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Only caller was the RtpFrameObject constructor, so it's
not needed in the interface.
To be able to delete downstream overrides, add a temporary
default implementation. Method will be completely deleted in part 2.
Bug: webrtc:9378
Change-Id: I9083b6284313b6ebce854c6f2cec4617953331d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112128
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25822}
Windows UWP allows an application to be built that targets
across all Windows 10 based systems and the Windows store.
Change-Id: I69694bb7e83fb01ad6db2438b065b55738cf01fd
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/110570
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25814}
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.
Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This is a follow up of a comment in
https://webrtc-review.googlesource.com/c/src/+/110105
It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video.
The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable.
Bug: webrtc:8789
Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458
Reviewed-on: https://webrtc-review.googlesource.com/c/110824
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25741}
This CL changes the behavior for RunFor and RunUntil so they do not
anymore restart the underlying streams every time they are called.
This has a side effect on the semantics of the calls. Previously,
both RunUntil and RunFor would restart the session and run until the
given time had passed. Now RunFor will still run for the provided
duration, however, to make the name of RunUntil more correct, it
will run until the time since start is equal to the max_duration
parameter. An extra overload of RunUntil was added to allow using
this behavior without providing an ending condition.
Bug: webrtc:9510
Change-Id: I9fe56a44116907fba3d102894b5c96af2ba6cffb
Reviewed-on: https://webrtc-review.googlesource.com/c/111502
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25726}
All implementations of ModifyReceiverCallConfig and
ModifySenderCallConfig configure the bitrate_config member only. So
replace these methods by ModifyReceiverBitrateConfig and
ModifySenderBitrateConfig.
This is a preparation for injecting RtpTransportControllerSend via
CallConfig. Then bitrates should be passed when constructing
RtpTransportControllerSend, and they can be deleted from CallConfig.
Bug: webrtc:7135
Change-Id: I6714158bd463dd485018713d0e26815919e5afcc
Reviewed-on: https://webrtc-review.googlesource.com/c/110780
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25624}
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).
Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.
Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
This prevents printing warning messages when the extensions aren't
found. The real parsing is done deeper in the stack and is unaffected.
Bug: webrtc:9510
Change-Id: Idf09f0e69c223bd4217be7044d21d1d0bbbdab92
Reviewed-on: https://webrtc-review.googlesource.com/c/110615
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25612}
Test that adapt down is triggered on overuse for different degradation preference configurations.
Bug: none
Change-Id: I326e979c10d09d17a7c1e6ece9a719f5fd4bff5f
Reviewed-on: https://webrtc-review.googlesource.com/c/97303
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25611}
Gain specified by fuzzer in APM config was too high.
Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25594}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This reverts commit 61c6e5643e.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.
Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.
To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.
Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
Additionally, AudioProcessing::GetStatistics(bool) is made pure
virtual and the default implementation in AudioProcessing is removed.
Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ
Bug: webrtc:9947, webrtc:8572
Change-Id: I123402bf7d6c49f3613154c469b818109d8fad43
Reviewed-on: https://webrtc-review.googlesource.com/c/108783
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25463}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
This makes the calculation more similar to the one in WebRTCVoiceEngine.
Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
Modified PressEnterToContinue() to run the Windows message loop in the
context of the SingleThreadedTaskQueueForTesting thread. The previous
PressEnterToContinue() was running the message loop in the context of
the main thread, but the "Local Preview" and "Loopback Video #0" are
created in the context of the SingleThreadedTaskQueueForTesting thread
and the message loop must be executed in the context of the thread that
created these windows in order for these windows to respond to any
event.
BUG=webrtc:9123
Change-Id: I2ec19f2569a940a510d3b2bd3881a89032d70332
Reviewed-on: https://webrtc-review.googlesource.com/c/67520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25408}
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705
Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
Rather fail at compile time than at run-time.
Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
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@{#25351}
Use absl::optional instead of special constant to show, that we failed
to get OutputPath in fileutils_override
Bug: webrtc:9792
Change-Id: Ice19a9bf425e88a747dd9b07e82dbb5bdc59685b
Reviewed-on: https://webrtc-review.googlesource.com/c/107630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25331}
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.
Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.
Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.
Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
This CL makes it possible to change transport routes while running
a scenario based test.
To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.
Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
This is a reland of 5ccdc1331f
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331f.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
We want sanitizer bots to show failure only for sanitizer defects.
To do so, this CL forces exit code to 0 unconditionally.
Sanitized binaries will turn it to 66 if there is any defect with diagnostic.
Bug: webrtc:9849
Change-Id: I46b683dcae12b76f1be177603af59e3f34bff3a9
Reviewed-on: https://webrtc-review.googlesource.com/c/107060
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25273}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
These two files were using absl::make_unique without #including the
header that declares it.
Bug: None
Change-Id: I03019c9a7e06370631680b474d04dd33716b0fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/107041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25266}
This is to avoid very long runs, resulting in time-outs.
NOTRY=True
Bug: chromium:895082
Change-Id: Iafdc3d10b3fb52f2d487547c954dca8ae7edb783
Reviewed-on: https://webrtc-review.googlesource.com/c/105960
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25175}
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.
Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
Also fix retransmission video send stream tests to not depend on actual frames sizes
Also, reduce key-frame scaling factor in FakeEncoder to better reflect real encoders behavior.
Bug: none
Change-Id: I33118160f3fec67ae8e732d9a85f0e9ee0784b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/105642
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25165}
Also renaming it Vp8TemporalLayers to show that it is codec specific.
Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
Avoid using post_encode_callback, instead add a new callback function
to ConfigurableFrameSizeEncoder. Intention is to delete
post_encode_callback and the EncodedFrameObserver class in a later cl.
Bug: None
Change-Id: I79c103adf11c8915878b3f7cacf24c9b02dd6373
Reviewed-on: https://webrtc-review.googlesource.com/c/104840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25064}
SetExecutablePath isn't used anymore.
Nobody was using the fancy select-per-platform functionality, and the
documentation was wrong anyway. In the cases somebody needed an
override per platform, they were using defines in their own test
instead. I think that is more verbose but more predictable and easy
to understand (see how it's done in audio_processing_unittest.cc
when loading output_data_mac, for instance).
Bug: webrtc:9792
Change-Id: I7289bf5883fe43852638922d7c7583eae0c08601
Reviewed-on: https://webrtc-review.googlesource.com/c/104482
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25062}
Add FakeVp8Decoder that parse width and height from the payload.
Add unit test for testing that width and height is set when decoding frames.
Bug: none
Change-Id: Ifbfff4f62f99625309ce0ef21cf89c76448769d8
Reviewed-on: https://webrtc-review.googlesource.com/c/103140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25038}
Fix the following issues with fuzz targets when built on Windows:
1. Fix audio_processing_fuzzer by making types match in
invocations of CheckedDivExact by explicitly casting to size_t.
2. Fix packet_buffer_fuzzer by including "frame_object.h" for
declaration of RtpFrameObject.
3. Fix rtcp_receiver_fuzzer by including "tmmb_item.h" for declaration
of TmmbItem.
Bug: chromium:891867
Change-Id: Iddc338360ca37d5fc31488ec908eb4cdb5cc7b94
Reviewed-on: https://webrtc-review.googlesource.com/c/103844
Commit-Queue: Jonathan Metzman <metzman@chromium.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25028}
This also moves the packet feedback tracking to RtpVideoSender.
Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
This code is much more sophisticated in that it doesn't rely
on argv[0], but rather asks the OS where our executable is.
We can then simply go two steps up since we count on running
in out/Whatever relative to the src dir. This is how Chromium
does it.
The aim here is to get rid of SetExecutablePath, which will
be the next CL.
Bug: webrtc:9792
Change-Id: I7da027b7391e759b1f612de12f27a244fe884c09
Reviewed-on: https://webrtc-review.googlesource.com/c/103121
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25017}
In https://webrtc-review.googlesource.com/c/src/+/102720 a new complex member
was added to VCMPacket. This member was overwritten with random data in the
fuzzer, which put it in an invalid state. To avoid that we save/restore it.
Bug: chromium:891597
Change-Id: I7b489afa727a028a542fbec55a4ee27ac54fa698
Reviewed-on: https://webrtc-review.googlesource.com/c/103462
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24972}
This refactoring merged PopulateCodecSpecific and FrameEncoded into a
single callback method. It also removes the FrameConfig parameter and
instead relies on the temporal layer to remember it internally.
Bug: webrtc:9012
Change-Id: I489b76821b534398ad452643f1322f411d3455b1
Reviewed-on: https://webrtc-review.googlesource.com/95681
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24957}
That is, cause a fatal error when a test produces such a result.
Bug: webrtc:9767
Change-Id: I588a34aa1e7e34b3036d5661e894676b21072862
Reviewed-on: https://webrtc-review.googlesource.com/c/101320
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24950}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.
Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
In the effort of enabling -Wglobal-constructors and
-Wexit-time-destructors, WebRTC has to remove the Winsock global
initializer.
This will also remove it from Chromium (since it was unused).
After this CL, applications will have to explicitly initialize Winsock
before using WebRTC, this can be done by using the class
rtc::WinsockInitializer provided in rtc_base/win32socketinit.h.
Bug: webrtc:9693, webrtc:9754
Change-Id: I4aae12ff43671ef2713a6fc4592e20759dc6b495
Reviewed-on: https://webrtc-review.googlesource.com/99660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24903}
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.
This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.
Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
This is a reland of 529d0d9795
Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
>
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
>
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}
Bug: webrtc:9106
Change-Id: I2eb894773b3f33ff6a980e8008e8248607e32668
Reviewed-on: https://webrtc-review.googlesource.com/102480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24882}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
Preparation for deleting EnableFrameRecordning, and also a step
towards landing of the new VideoStreamDecoder.
Bug: webrtc:9106
Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
Reviewed-on: https://webrtc-review.googlesource.com/97660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24861}
The old pointer-to-submodule interface is replaced with
AudioProcessing::Config settings.
Bug: webrtc:9535
Change-Id: I5580d690fdd7664f48fa274b39f12cc41f69da37
Reviewed-on: https://webrtc-review.googlesource.com/102020
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24847}
This reverts commit 3f4a4fad8c.
Reason for revert: Breaking internal tests
Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
>
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
>
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.
Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
to satisfy a stricter check introduced in
503174a3e1
The file is supposed to contain actual gtest results, so having an
empty one is a workaround, but this just returns things to the way
they were.
TBR: phoglund@webrtc.org
No-Try: True
Bug: webrtc:9767, chromium:885194
Change-Id: I693cc2df9dfcafd7b728deb9efd445d8fe2c4edf
Reviewed-on: https://webrtc-review.googlesource.com/101301
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24783}
When WebRTC fuzzers tests are built on Chromium bots they need to link
with Chromium's implementation of metrics.
TBR=phoglund@webrtc.org
Bug: webrtc:9631
Change-Id: I1c955e646366b6b37d3ca595888e8cc94fe1b00e
Reviewed-on: https://webrtc-review.googlesource.com/100940
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24771}
This component was added to work around an issue in iOS 8, which is
no longer supported by WebRTC. It's removal is made more urgent by
the fact that it prevents WebRTC being used by iOS extensions.
Bug: webrtc:9335
Change-Id: I2a3327534fe6d5014c34a9e908096d825e8149e3
Reviewed-on: https://webrtc-review.googlesource.com/87822
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24688}
Fix race between FrameGeneratorCapturer destructor and inserting a frame.
(aka do not use this after reposting self to a TaskQueue)
Separate periodic case from one-time case.
When Generator can't keep up with fps, drop frames instead of trying to catch up.
Use absl::make_unique and absl::WrapUnique
Bug: None
Change-Id: I9d5d1fcacf174e28d83310099b79e26ece9b7b37
Reviewed-on: https://webrtc-review.googlesource.com/98844
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24654}
This is a reland of 711a31aead
Changes since original landing:
Rename methods only used by tests, mainly via FakeClock,
MessageQueueManager::ProcessAllMessageQueues
--> ProcessAllMessageQueuesForTesting
MessageQueue::IsProcessingMessages
--> IsProcessingMessagesForTesting
Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().
Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().
Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}
Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
This is a reland of da0898dfae
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
This CL adds the utilities to generate and parse mDNS messages (RFC 1035
and RFC 6762).
TBR=phoglund@webrtc.org
Bug: webrtc:9605
Change-Id: Id6121c17926887cd3a41a2dfc829462fd15f3a4c
Reviewed-on: https://webrtc-review.googlesource.com/93241
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24505}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.
Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
This is reland after fixes for broken perf tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/91861
Bug: none
Change-Id: I6999a499408787be43a74a26a16b7826a0814a7b
Reviewed-on: https://webrtc-review.googlesource.com/95182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24383}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.
Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
There is no difference between how we handle "generic" and "unkown" codecs,
so we don't need to represent both.
Bug: webrtc:8136
Change-Id: I42b0dbc8a0bae67cc21742303c963c8dd5bde1f6
Reviewed-on: https://webrtc-review.googlesource.com/92086
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24316}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
Since the linux_memcheck trybot is no more, this CL removes all the
code needed to make it work.
Bug: webrtc:7737, webrtc:8356, webrtc:9570
Change-Id: I09a9467b8bf895146a3384c2c915b54662721af6
Reviewed-on: https://webrtc-review.googlesource.com/90863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24201}
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.
This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org
Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
This reverts commit 771b50ca0b.
Reason for revert: Introduces error-prone config.
Original change's description:
> Add one-stop-shop for built-in AEC toggling in APM
>
> This does not change what AEC functionality is available.
> However, a client that only uses this interface - and not the submodule
> pointer accessors - gets simpler code, and is guaranteed not to run any
> two AECs in tandem.
>
> The submodule interface EchoControlMobile is being deprecated in
> https://webrtc-review.googlesource.com/c/src/+/89392
>
> Bug: webrtc:9535
> Change-Id: Id9326074e566be6d8768010fc421c457beff402c
> Reviewed-on: https://webrtc-review.googlesource.com/89386
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24066}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I43283a1b22538a4caa77313499989146b2ce67f1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/90060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24067}
This does not change what AEC functionality is available.
However, a client that only uses this interface - and not the submodule
pointer accessors - gets simpler code, and is guaranteed not to run any
two AECs in tandem.
The submodule interface EchoControlMobile is being deprecated in
https://webrtc-review.googlesource.com/c/src/+/89392
Bug: webrtc:9535
Change-Id: Id9326074e566be6d8768010fc421c457beff402c
Reviewed-on: https://webrtc-review.googlesource.com/89386
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24066}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6d3b45de9dca3a5a04f0cdd5583919d35a585a7e
Reviewed-on: https://webrtc-review.googlesource.com/89043
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24018}
Previously, the fuzzer read a int16_t and converted to float. That is
how float audio samples were generated. This CL changes the fuzzer to
read floats directly, and then sanitize them.
Bug: webrtc:7820
Change-Id: Icc526611466c10dd4222b19a4d4b4fd26643812a
Reviewed-on: https://webrtc-review.googlesource.com/85343
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24001}
The fuzzer figured out that 3 bytes is enough to fuzz a package.
2 bytes for packet length, and 1 byte of actual packet. A 20K test case
can generate > 6000 packets. It does not seem like efficient fuzzing.
This CL simply stops execution when 200 packets have been generated.
That corresponds to 4 seconds of 20 ms packets.
Bug: chromium:840115
Change-Id: Id2742a6f8021134bacd8a6e8c71b32f20c7f1086
Reviewed-on: https://webrtc-review.googlesource.com/88566
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24000}
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.
Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
This ensures the event logs in CallTest will be used by default.
Bug: webrtc:9510
Change-Id: I9df82b5ef39e1b2cba2789f8c5c7a9fed3c4c2f6
Reviewed-on: https://webrtc-review.googlesource.com/88562
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23970}
This prepares for replacing single instance members with vectors in a
follow up CL.
Bug: webrtc:9510
Change-Id: Ie05436ec89a0af9ce9fe9cece9842a39227246ec
Reviewed-on: https://webrtc-review.googlesource.com/88180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23968}
This reduces code duplication. FunctionVideoEncoderFactory is modified
to allow providing a function that takes an argument for the format.
Bug: webrtc:9510
Change-Id: I67fee84af4968a51326b52db35f3eb0c65848735
Reviewed-on: https://webrtc-review.googlesource.com/88222
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23965}
This makes it possible to use them in VideoQualityTest and prepares for
allowing saving logs in other tests as well.
Bug: webrtc:9510
Change-Id: I3b1cc187d88e4f17745414433c2f96efd836a302
Reviewed-on: https://webrtc-review.googlesource.com/88561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23964}
There are many new killswitches for AEC3 code.
By fuzzing them, we ensure the code is more thoroughly tested in case
we need to trigger any switches.
Bug: webrtc:9413
Change-Id: I0bc7609e4d71fc820abfbba7d481b2374c3587cb
Reviewed-on: https://webrtc-review.googlesource.com/88225
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23960}
Also prepared it for comming changes to RTPVideoHeader.
Bug: none
Change-Id: I60e57f779aee4ee98b6eec3d6631e43ac682c37f
Reviewed-on: https://webrtc-review.googlesource.com/87825
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23911}
The size limit is set to correspond to approximately 5 seconds of
decoded audio at the codecs' normal operating bitrates. This is to
avoid timeouts on the bots.
NOTRY=true
Bug: chromium:840115
Change-Id: I74b3c196259e03981aa2c4ef349e6e1334e9bf58
Reviewed-on: https://webrtc-review.googlesource.com/87302
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23857}
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant.
Bug: none
Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98
Reviewed-on: https://webrtc-review.googlesource.com/87261
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23856}
Add support for stereo in APM fuzzer.
Bug: webrtc:9413
Change-Id: Ie4b19037bd4613c050b03ad0bacf0f44f9feccd3
Reviewed-on: https://webrtc-review.googlesource.com/87221
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23854}
This calls webrtc::AudioProcessing::set_stream_key_pressed, which
opens up a lot of code paths in the transient suppressor.
The change breaks historical fuzzer test cases.
Bug: webrtc:9413
Change-Id: I1f593a98286c7e7c0fc6751d16df40ad813dbd70
Reviewed-on: https://webrtc-review.googlesource.com/86950
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23840}
This is the preferred way.
NOTRY=True
Bug: none
Change-Id: I305d4d9cb7b66e01427958e1416d672badd72af0
Reviewed-on: https://webrtc-review.googlesource.com/86948
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23837}
This avoids fuzzer timeouts on the bot.
NOTRY=True
Bug: chromium:857404
Change-Id: I480c53f005536029c667b9f41aab3ecaca14d125
Reviewed-on: https://webrtc-review.googlesource.com/86945
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23833}
This is a reland of 80c4cca491
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
to list of extensions supported by RtpPacket.
Bug: webrtc:9361
Change-Id: Iabee824381be3776e17e95f32507058607fc0bc8
Reviewed-on: https://webrtc-review.googlesource.com/85346
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23788}
This slightly increases fuzzer coverage of the APM.
(.25 % points more line coverage.)
Bug: webrtc:9413
Change-Id: Ic992423f1dcf34fa0aa9649c8035a8e48b0ccdb2
Reviewed-on: https://webrtc-review.googlesource.com/85342
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23732}
Added distinction between number of configured and number of actively
encoded spatial layers and include number of actively encoded spatial
layers in ssData. Modified layer_filtering_transport.cc test to
parse from the RTP header and use the number of actively encoded
spatial layers for filtering spatial video layers.
Bug: webrtc:9425
Change-Id: Ic9f8895ab08b0626f9bb53a75ec33d8e7eb8706e
Reviewed-on: https://webrtc-review.googlesource.com/84243
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23716}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
This reverts commit 07efe436c9.
Reason for revert: Breaks downstream project.
cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).
Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
>
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
> under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
>
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}
TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com
Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
Fuzzes the config and audio inputs to GainControlImpl.
Seems able to cover a few hundred lines of code that the APM fuzzer hasn't been able to reach.
Bug: webrtc:9413
Change-Id: I32776505be9c416ec03113c12437a92dcfadd827
Reviewed-on: https://webrtc-review.googlesource.com/84589
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23709}
This reverts commit 80c4cca491.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
This CL removes the remaining beamformer parts from the APM.
Bug: webrtc:9402
Change-Id: I9ab2795bd2813d17166ed0925125257b82d98a74
Reviewed-on: https://webrtc-review.googlesource.com/83340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23694}
Only user was the replay.cc tool, when dumping frames to a file. It is
changed to instead inject a special decoder.
Bug: None
Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f
Reviewed-on: https://webrtc-review.googlesource.com/83121
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23675}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
The fuzzer data is used to configure the field trials of the AEC.
This increases fuzzer coverage of modules/audio_processing/aec3/ by roughly 500 lines of code, ~ 3 % points increase in APM coverage for desktop Chrome.
Bug: webrtc:9413
Change-Id: Iea9059747a8492a7ca2091a359e7883750c45b27
Reviewed-on: https://webrtc-review.googlesource.com/83732
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23650}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'test rtc_tools'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ibb43c737f4c45fe300736382b0dd2d8ab32c6377
Reviewed-on: https://webrtc-review.googlesource.com/83944
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23642}
In rare case the packets number may loop around and in the same FEC-protected group the packet sequence number became out of order.
Bug: chromium:850493
Change-Id: Ice82aafd537e0edc1dbdb8b934e11e7c42a4cf60
Reviewed-on: https://webrtc-review.googlesource.com/82802
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23633}
The echo detector is currently stored as a unique_ptr, but when injecting an echo detector, a scoped_refptr makes more sense since the ownership will be shared.
Bug: webrtc:8732
Change-Id: I2180014acb84f1cd5c361864a444b7b6574520f5
Reviewed-on: https://webrtc-review.googlesource.com/83325
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23610}
And update most internal calls to use it.
Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.
Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
This limits the SDP to 16KB, which sounds enough.
Bug: chromium:813328
Change-Id: I58c7b3e073108fd7b3495e8182b5c632e9619fe7
Reviewed-on: https://webrtc-review.googlesource.com/78280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23360}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".
It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.
Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
Vp9 encoder supports several inter-layer prediction modes. This adds
possibility to control and test them in video/ss/sv loopback.
Filtering of sent packets has been modified. In addition to high
spatial and temporal layers it now filters out packets of low spatial
layers where non_ref_for_inter_layer_pred bit is set to true.
Bug: none
Change-Id: I17b1ee8f1ac1d70a6914eb86d153790ef2da9679
Reviewed-on: https://webrtc-review.googlesource.com/76540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23233}
The previous effort of building WebRTC without SW codecs stopped when
libjingle_peerconnection was possible to build. In order to make the
group("default") target build, this basically updates a bunch of
tests to explicitly depend on the built-in software video codecs.
Bug: webrtc:7925
Change-Id: I2715414770c197fca01cb8dbde173a21f4434500
Reviewed-on: https://webrtc-review.googlesource.com/70503
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23216}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
The fuzzer is very simple. It only considers the default encoder
configuration at this point.
Bug: chromium:826914
Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a
Reviewed-on: https://webrtc-review.googlesource.com/75261
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23192}
Intend to delete in a later cl.
Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
This CL moves the responsibility for demuxing from FakeNetworkPipe
to DirectTransport. This makes the interface for FakeNetworkPipe more
consistent. It exposes fewer different interfaces for different usages.
It also means that any time degradations applied to the packets due in
FakeNetworkPipe in tests will now be propagated to Call in a more
realistic manner. Previously the time was set to uninitialized which
meant that Call filled in values based on the system clock.
Bug: webrtc:9054
Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0
Reviewed-on: https://webrtc-review.googlesource.com/64260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23017}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
This is a reland of fb82fcc7f9
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This reverts commit fb82fcc7f9.
Reason for revert: Appears to break Chromium, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Linux%20Tester/43756, where remoting_unittests failed.
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: I47ee3ac42e62472d825a08c98e28f9ae53ec9fff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/70600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22914}
This CL includes the changes from this CL:
https://webrtc-review.googlesource.com/c/src/+/63642
Bug: webrtc:8955
Change-Id: If95cdec59f25e97c6ff5ea45a52d6113128a0921
Reviewed-on: https://webrtc-review.googlesource.com/64822
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22910}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
This reverts commit aaa85ae565.
Reason for revert: Breaks iOS64 Debug trybot: https://uberchromegw.corp.google.com/i/internal.client.webrtc/builders/iOS64%20Debug/builds/14014
The failure being at:
../../test/fpe_observer_unittest.cc:93: Failure
Expected equality of these values:
0x009f
Which is: 159
all_flags
Which is: 31
It looks like the missing flag may be "FE_FLUSHTOZERO"?
Original change's description:
> Reland "Floating-point exception observer for unit tests"
>
> This reverts commit e3d522dd6b.
>
> Reason for revert: Disabling test failing in downstream projects.
>
> Original change's description:
> > Revert "Floating-point exception observer for unit tests"
> >
> > This reverts commit 3fb3939896.
> >
> > Reason for revert: Downstream projects failures.
> >
> > Original change's description:
> > > Floating-point exception observer for unit tests
> > >
> > > This CL adds a simple tool that let a unit test fail if a floating
> > > point exception occurs. It is possible to focus on specific exceptions.
> > > Note that FloatingPointExceptionObserver is only effective in debug
> > > mode. For this reason, the related unit tests only run in debug mode.
> > > Plus, due to some platform-specific limitations, not all the floating
> > > point exceptions are available on Android.
> > >
> > > Bug: webrtc:8948
> > > Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> > > Reviewed-on: https://webrtc-review.googlesource.com/58097
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22768}
> >
> > TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
> >
> > Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8948
> > Reviewed-on: https://webrtc-review.googlesource.com/67380
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22769}
>
> TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8948
> Change-Id: I7584d941b227277a271323b47bc70945af999758
> Reviewed-on: https://webrtc-review.googlesource.com/69060
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22848}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8948
Change-Id: Ia377cea165211a0fad8f7ab29baae3eee64395c3
Reviewed-on: https://webrtc-review.googlesource.com/70280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22886}
This reverts commit e3d522dd6b.
Reason for revert: Disabling test failing in downstream projects.
Original change's description:
> Revert "Floating-point exception observer for unit tests"
>
> This reverts commit 3fb3939896.
>
> Reason for revert: Downstream projects failures.
>
> Original change's description:
> > Floating-point exception observer for unit tests
> >
> > This CL adds a simple tool that let a unit test fail if a floating
> > point exception occurs. It is possible to focus on specific exceptions.
> > Note that FloatingPointExceptionObserver is only effective in debug
> > mode. For this reason, the related unit tests only run in debug mode.
> > Plus, due to some platform-specific limitations, not all the floating
> > point exceptions are available on Android.
> >
> > Bug: webrtc:8948
> > Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> > Reviewed-on: https://webrtc-review.googlesource.com/58097
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22768}
>
> TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8948
> Reviewed-on: https://webrtc-review.googlesource.com/67380
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22769}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8948
Change-Id: I7584d941b227277a271323b47bc70945af999758
Reviewed-on: https://webrtc-review.googlesource.com/69060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22848}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
A part of the Audio Processing Module interface is GetStatistics. The
call collects stats from submodules. We make sure these calls are made
by the fuzzer to cover that code path.
Bug: webrtc:7820
Change-Id: Ia8f89d9838602dcb2599f676bd5c43e815bbf791
Reviewed-on: https://webrtc-review.googlesource.com/68980
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22817}
When building test:test_support_unittests with is_official_build=true,
the linker fails with the following error:
duplicate symbol: webrtc::videocapturemodule::VideoCaptureImpl::Create(
char const*)
>>> defined in obj/modules/video_capture/video_capture_internal_impl/\
video_capture_linux.o
>>> defined in obj/modules/video_capture/libvideo_capture.a(\
video_capture_external.o)
After looking at both test:test_support_unittests and test:test_support,
it seems these targets had unused dependenicies. This CL removes them
and fixes the duplicated symbol error.
The GN flag is_official_build changes some configurations down in the
toolchain, that is probably why building with is_official_build=false
was not triggering the problem.
In any case, build targets in test/ need to be cleaned up because they
depend on too many things.
Bug: webrtc:9117
Change-Id: Icfdae3b5610f1c873ccdd0292c12ef946dea79af
Reviewed-on: https://webrtc-review.googlesource.com/67161
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22789}
This reverts commit 3fb3939896.
Reason for revert: Downstream projects failures.
Original change's description:
> Floating-point exception observer for unit tests
>
> This CL adds a simple tool that let a unit test fail if a floating
> point exception occurs. It is possible to focus on specific exceptions.
> Note that FloatingPointExceptionObserver is only effective in debug
> mode. For this reason, the related unit tests only run in debug mode.
> Plus, due to some platform-specific limitations, not all the floating
> point exceptions are available on Android.
>
> Bug: webrtc:8948
> Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> Reviewed-on: https://webrtc-review.googlesource.com/58097
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22768}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8948
Reviewed-on: https://webrtc-review.googlesource.com/67380
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22769}
This CL adds a simple tool that let a unit test fail if a floating
point exception occurs. It is possible to focus on specific exceptions.
Note that FloatingPointExceptionObserver is only effective in debug
mode. For this reason, the related unit tests only run in debug mode.
Plus, due to some platform-specific limitations, not all the floating
point exceptions are available on Android.
Bug: webrtc:8948
Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
Reviewed-on: https://webrtc-review.googlesource.com/58097
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22768}
Added functionality on the FakeNetworkPipe to introduce arbitrary
clock offsets. This offset is added to the reported receive time of
all packets. This prepares for a later CL using this to test correction
of receive time stamps.
Bug: webrtc:9054
Change-Id: I811b3aa8359bc917f59443088d8a418368242db9
Reviewed-on: https://webrtc-review.googlesource.com/64726
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22763}
This reverts commit 04dd176862.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
This is a reland of bc900cb1d1
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
This reverts commit bc900cb1d1.
Reason for revert: Broke downstream projects.
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.
EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.
The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.
Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
We currently have one build target containing everything for audio_device: the interfaces,
the "fine" audio buffer, and the actual implementations for each platform.
Since we are planning to move the Android implementation to the sdk/android folder,
we only want to depend on the interfaces and the "fine" audio buffer, not the other platform
specific implementations. This CL splits the audio_device target into three different targets:
the interfaces, the fine audio buffer, and the platform specific implementations. The default
audio_device target now points to the interfaces instead.
Bug: webrtc:7452
Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd
Reviewed-on: https://webrtc-review.googlesource.com/61321
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22452}
Why this dep is here is lost to history. Everything works
without it though.
Bug: webrtc:8821
Change-Id: Ie0d763fb8a6508f7177a2f4bc9b7d909b9b02eb6
Reviewed-on: https://webrtc-review.googlesource.com/61962
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22441}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
TBR=stefan@webrtc.org, philipel@webrtc.org
Originally reviewed on: https://webrtc-review.googlesource.com/33013
Bug: webrtc:8910
Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0
Reviewed-on: https://webrtc-review.googlesource.com/61782
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22430}
This reverts commit 31a12c557d.
Reason for revert: Breaks downstream project.
Original change's description:
> Add ability to emulate degraded network in Call via field trial
>
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
>
> Also includes some refactorings.
>
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
Bug: webrtc:8910
Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
Reviewed-on: https://webrtc-review.googlesource.com/33013
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22418}
This is a reland of d90a7e8424
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Ib6e37ce4bc0bae903dd72f49ffdc2ee583d75491
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/61120
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22376}
The number of calls to ComfortNoiseDecoder::Generate() was determined
by the fuzzer input, and was chosen between 0 and 255. This would
sometimes lead to very long runs, with questionable merit. With this
change, the number of call to Generate() is limited to 17 (an
arbitrary small integer).
Bug: chromium:820078
Change-Id: I27b5c7f0b72d53370d002a6b157d4451079a0ba9
Reviewed-on: https://webrtc-review.googlesource.com/60941
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22360}
Previously, the only user of this code was the
VideoProcessorIntegrationTest. We have now changed that
test to directly calculate image quality metrics using libyuv,
similar to how the full stack tests and browser tests work.
Bug: webrtc:8448
Change-Id: Ia7a607d7ddc37741fba76d56aa7297851ffa1c6b
Reviewed-on: https://webrtc-review.googlesource.com/43760
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22341}
This is a reland of d90a7e8424
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Iba2e89aee73a73a0372edea26933d6a7ea2e0ec9
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/60600
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22336}
This reverts commit d90a7e8424.
Reason for revert:
Fails on Win ASan bots.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc%2FWin32_ASan%2F4002%2F%2B%2Frecipes%2Fsteps%2Fvideo_engine_tests%2F0%2Fstdout
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
TBR=phoglund@webrtc.org,sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: If6bfdd42556517db0dd6bda01f5d3d901ff56b0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/60560
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22332}
This CL adds two new tests to perf, covering I420 and I420A input to multiplex
codec. In order to have the correct input, it adds I420A case to
SquareGenerator and corresponding PSNR and SSIM calculations.
Bug: webrtc:7671
Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
Reviewed-on: https://webrtc-review.googlesource.com/52180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22330}
So that we can avoid dependency cycles.
Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. It will be done in 3 steps:
1. Test/fake_audio_device will be moved to production part of WebRTC
source code and renamed to test_audio_device_module. Old header is
replaced with alias to the new one.
2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice.
3. test/fake_audio_device will be removed.
This CL implements 1st step.
Bug: webrtc:8946
Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f
Reviewed-on: https://webrtc-review.googlesource.com/59740
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22325}
It holds the same information as codecType, but in different format.
Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
This reverts commit 8ea5f9ae5b.
Reason for revert: breaks downstream project
Original change's description:
> Separate test/fake_audio_device on API and implementation.
>
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
>
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
>
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}
TBR=kwiberg@webrtc.org,titovartem@webrtc.org
Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.
Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
We found out that
int16_t x = test::FuzzDataHelper::ReadOrDefaultValue(0)
reads 4 bytes from the fuzzer input instead of 2. That means that
almost half the bits in the input data to audio_processing_fuzzer are
ignored. This change adds template arguments to force reading 2 bytes
when we only need 2.
We also add a small manually generated corpus. During local testing we
let the fuzzer run for a few hours on an empty corpus. Adding the
manually-generated files resulted in an immediate coverage increase by
~3%, and then by another 3% over the next few hours.
The manually generated corpus contains a short segment of speech with
real echo. We suspect that triggering Voice Activity Detection or echo
estimation filter convergence can be difficult for an automatic
fuzzer.
We remove the Level Controller config. We read 20 bytes extra after the
config to guard against future configuration changes.
Bug: webrtc:7820
Change-Id: If60c04f53b27c519c349a40bd13664eef7999368
Reviewed-on: https://webrtc-review.googlesource.com/58744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22269}
The additional include is needed in order to use EXPECT_NONFATAL_FAILURE()
in unit tests.
Bug: webrtc:8948
Change-Id: If5b9ceb89a3a36480657d094cfabc81c9b0e15b7
Reviewed-on: https://webrtc-review.googlesource.com/58096
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22227}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
The length of the fuzzer input can sometimes be really long (more than
1000000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 200000 bytes.
NOTRY=TRUE
Bug: chromium:802149
Change-Id: Ia9d2f080602bba8ff70c5f0575bb9ecfa99c537c
Reviewed-on: https://webrtc-review.googlesource.com/57581
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22183}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802193
Change-Id: Id32174611fadb480f4e2c6b4f553a2ba0fa5b493
Reviewed-on: https://webrtc-review.googlesource.com/57580
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22182}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802245
Change-Id: Ibe02b6de932d900408f870d9ba440b7b8e08dc0e
Reviewed-on: https://webrtc-review.googlesource.com/57180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22181}
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.
With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.
Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
new is an unsafe construct, while these specific cases were properly
handled it is a code smell and using unique_ptr from the start makes the
code more obviously correct.
Bug: None
Change-Id: I2554cef8d3a8432a3ced1623292fae0adff9421d
Reviewed-on: https://webrtc-review.googlesource.com/56620
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22147}
MockAudioEncoder was calling a mocked Die function on itself in its
destructor. This outputs "Uninteresting mock function call" warning if
the Die call was not expected. This is true even if a NiceMock is used
to suppress the warnings.
The purpose of testing that the destructor is called might be to protect
against memory leaks when audio encoder ownership is transferred using a
raw pointer. However, this case is already covered by msan checks.
Bug: None
Change-Id: I0603c417b4b239027859228e05ebcf83ff5aaf18
Reviewed-on: https://webrtc-review.googlesource.com/56183
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22146}
Since rtp transport controller send owns the congestion controller it
also should own the bitrate configuration logic, this way it can
initialize the send side congestion controller with the bitrate
configuration.
Bug: webrtc:8415
Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553
Reviewed-on: https://webrtc-review.googlesource.com/54303
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22127}
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.
The other IGC and LE submodules were added in previous CLs [1] and
[2].
This CL also turns on AGC2 in the APM fuzzer.
[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381
Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
Deleting the apparently unused include of api/rtp_headers from
common/video/include/video_frame.h broke the PayloadRouter and
VideoSendStream code under video/. Missing declaration of the
RtpPayloadState struct declared in api/rtp_headers.h. Moving the
declaration of that struct to payload_router.h (outside of the api),
since it's used only internally in video/, and that seemed to be a
more logical place for it.
Bug: webrtc:7504
Change-Id: Ibed8233dfeea8bdf144db5422cdf897da824d6ee
Reviewed-on: https://webrtc-review.googlesource.com/53701
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22080}
When we run webrtc_perf_tests with gtest-parallel, each test is run
individually, and this results in the file with the perf results being
overwritten each time.
To avoid this, we won't use gtest-parallel when running webrtc_perf_tests,
so we will simply run the binary directly.
TBR=phoglund@chromium.org
Bug: chromium:755660
Change-Id: I24db36e512fcf604a3de2adf4d0b4325b2c3d1ae
Reviewed-on: https://webrtc-review.googlesource.com/49340
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21982}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
Rtp Packets in webrtc expected to be less that 1500,
i.e. way less that 2^16 bytes for extensions block.
This CL explicitly discards longer extension.
Bug: chromium:809046
Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734
Reviewed-on: https://webrtc-review.googlesource.com/48061
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21910}
When a simulcast stream is enabled or disabled, we want this state
change to be reflected properly in the RtpRtcp modules. Each video send
stream can contain multiple rtp_rtcp_modules pertaining to different
simulcast streams. These modules are currently all turned on/off when
the send stream is started and stopped. This change allows for
individual modules to be turned on/off. This means if a module stops
sending it will send a bye message, so the receiving side will not
expect more frames to be sent when the stream is inactive and the
encoder is no longer encoding/sending images.
Bug: webrtc:8653
Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4
Reviewed-on: https://webrtc-review.googlesource.com/42841
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21880}
This is a reland of 001546da95
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
This is done to solve a problem where a string literal is implicitly cast
to a temporary std::string when calling webrtc::test::InitFieldTrialsFromString
which passes a pointer to the internal representation to
webrtc::field_trial::InitFieldTrialFromString(char*). This pointer is
stored for later use, but the temporary std::string is destroyed as soon
as the function returns.
Using webrtc::field_trial::InitFieldTrialFromString(char*) instead,
avoids the implicit casts (but the caller still needs to ensure that
the char* outlives the program). The validation previously done by
webrtc::test::InitFieldTrialsFromString can now be done by manually
calling webrtc::test::ValidateFieldTrialsStringOrDie(const std::string&).
Add system_wrappers:field_trial_default as a direct dependency to
various targets to allow including the field_trials_default.h header.
Bug: webrtc:8812
Change-Id: Ib5a641ea255b1c16a8f7f35e1fe67f6c38a61da6
Reviewed-on: https://webrtc-review.googlesource.com/46141
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21856}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
This is a reland of bbdabe50db.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
Delete the subdirectory modules/media_file, and all references to it.
Bug: none
Change-Id: I19d86420a7d1d51cb6174c914a90484918106c5a
Reviewed-on: https://webrtc-review.googlesource.com/40540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21790}
This reverts commit 4954a77cf8.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
This is a reland of bbdabe50db.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
This reverts commit bbdabe50db.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
This feature is not needed in video codec testing framework. In WebRTC
video codecs never deal with packet loss. Packet loss is handled by
jitter buffer which prevents passing of incomplete frames to decoder.
Bug: webrtc:8768
Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6
Reviewed-on: https://webrtc-review.googlesource.com/40740
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21722}
Prior to this change, in certain circumstances the RTP header length
used when creating a RedPacket was incorrect. This was due to an
assumption that a new media packet would _always_ be added to the
UlpfecGenerator's internal media packet buffer. This is not correct,
and the fix is to keep track of whatever RTP header length that is
currently correct.
Bug: webrtc:8767
Change-Id: I6d61429a19d4693dde9330f0469d13c5dfbeac52
Reviewed-on: https://webrtc-review.googlesource.com/40600
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21720}
This is a reland of 18c4261339
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=sprang@webrtc.org,stefan@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8630
Change-Id: Ib3df6f9b7158bff362a7ec66fc57e368682c5846
Reviewed-on: https://webrtc-review.googlesource.com/40980
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21688}
This is a reland of 1880c7162b
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
This reverts commit 1880c7162b.
Reason for revert: breaks internal tests
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
- Run analysis after all frames are processed. Before part of it was
done at bitrate change points;
- Analysis is done for whole stream as well as for each rate update
interval;
- Changed units from number of frames to time units for some metrics
and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
'time to reach target bitrate, sec';
- Changed data type of FrameStatistic::max_nalu_length (renamed to
max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
use such advanced data type in such low level data structure.
Bug: webrtc:8524
Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
Reviewed-on: https://webrtc-review.googlesource.com/31901
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21653}
This reverts commit 18c4261339.
Reason for revert: Broke internal tests
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,shampson@webrtc.org
Change-Id: I0aeb743cbd2e8d564aa732c937587c25a4c49b09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/39883
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21647}
Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
Bug: webrtc:8653
Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
Reviewed-on: https://webrtc-review.googlesource.com/37740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21646}
Going through the coverage of audio_processing_fuzzer, it was noticed
that it didn't cover AEC-m and AEC-2 code. Therefore this CL adds 2
fuzzer targets that only fuzz the previous generation echo cancellers.
To avoid code duplication, the APM running code was broken out in a
new GN target. We have also changed all fuzzing code to use the
FuzzDataHelper class to avoid manual pointer arithmetic.
Bug: webrtc:7820
Change-Id: Ifea3266e396b487952a736945577fccea15d0e01
Reviewed-on: https://webrtc-review.googlesource.com/36500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21638}
aleloi has read the Chromium fuzzer guides and covered lots of APM
code with fuzzing tests. He'd like to share responsibility for making
fuzzers faster and have high coverage.
Bug: None
NOTRY: True
Change-Id: I45db63349ca9d4432ebc69ed3c84ec2fc0f3f227
Reviewed-on: https://webrtc-review.googlesource.com/39923
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21637}
This method was unused. When deleted, also configuration of
receive-side RTP header extensions in this class becomes unused.
Header extensions are parsed in Call.
Bug: None
Change-Id: Iad76abf72962f3d91e85dde43541c3b6a9522b7e
Reviewed-on: https://webrtc-review.googlesource.com/39782
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21636}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split
Bug: None
Change-Id: Ifca0183c2675a8d4e74c21b86f1724d9b2c01f5d
Reviewed-on: https://webrtc-review.googlesource.com/23579
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21580}
* VoEBase contains only stub methods (until downstream code is
updated).
* voe::Channel and ChannelProxy classes remain, but are now created
internally to the streams. As a result,
internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
for testing.
* Stream classes share Call::module_process_thread_ for their RtpRtcp
modules, rather than using a separate thread shared only among audio
streams.
* voe::Channel instances use Call::worker_queue_ for encoding packets,
rather than having a separate queue for audio (send) streams.
Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}
This reverts commit c73e1f4378.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.
Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
This is a reland of d2b912aed1
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Bug: webrtc:8630
Change-Id: I7429d9e270c9ecb4dfaf6aef85d3055c47658631
Reviewed-on: https://webrtc-review.googlesource.com/35600
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21524}
This adds a SetCapturer function to testing::FakeAudioDevice::PulsedNoiseCapturer
that can be used to update the volume of the generated audio mid-call. It also modifies
CreatePulsedNoiseCapturer to use PulsedNoiseCapturer's type directly so that its new
function is visible for the callers.
Bug: webrtc:8666
Change-Id: I47726e242ccf221f5511e2797b2954ce035ba371
Reviewed-on: https://webrtc-review.googlesource.com/34650
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21521}
This will require a manual roll to downstream projects, since
the //test:perf_test target was introduced.
This is a reland of 10a8e7a9b5
Original change's description:
> iOS: Save perf results under Documents/perf_result.json
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:7156
> Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5
> Reviewed-on: https://webrtc-review.googlesource.com/29202
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21244}
R=henrika@webrtc.org, phoglund@webrtc.org
Bug: webrtc:7156
Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424
Reviewed-on: https://webrtc-review.googlesource.com/37643
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21518}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
VideoSourceInterface is clearly an integral part of
mediastreaminterface.h already, so moving that interface to api makes
sense. This also resolves a circular dependency in call/.
Bug: webrc:6828
Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c
Reviewed-on: https://webrtc-review.googlesource.com/37500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21498}
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
Desktop capturing on Linux will be disabled in this case, but everything
can be built without any X11 development libraries installed.
BUG=webrtc:5716,webrtc:8319
Change-Id: I01bd6a4b02816b407be19476e22ff073d264b496
Reviewed-on: https://webrtc-review.googlesource.com/32360
Reviewed-by: Henrik Andreassson (OOO until Jan 2) <henrika@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21462}
This reverts commit d2b912aed1.
Reason for revert: broke internal tests
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,pthatcher@webrtc.org,shampson@webrtc.org
Change-Id: If82810072e21818ae452a0fc3f984d44e5dac70c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8630
Reviewed-on: https://webrtc-review.googlesource.com/35540
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21399}
I followed the wiring path for the max bitrate.
Doc:
https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
Bug: webrtc:8630
Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
Reviewed-on: https://webrtc-review.googlesource.com/30380
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21397}
The functionality is moved into AudioState.
TBR: henrika@webrtc.org
Bug: webrtc:4690
Change-Id: I015482ad18a39609634f6ead9e991d5210107f0f
Reviewed-on: https://webrtc-review.googlesource.com/34502
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21338}
The functionality is moved into AudioState.
Bug: webrtc:4690
Change-Id: Iee1bfd185566c9507422e8eea8a2cce02baaefe1
Reviewed-on: https://webrtc-review.googlesource.com/33521
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21324}
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This reverts commit f711428898.
Reason for revert: Breaks downstream projects.
Original change's description:
> Use std::fstream instead of rtc::File to write perf results + rename flag.
>
> Use std::fstream instead of rtc::File to write perf results.
> On Android, when I use rtc::File, the results are not written for some reason.
>
> Also rename the flag to '--chartjson_result_file'.
>
> Bug: webrtc:8566
> Change-Id: I32215e2233e18690c41050dfd35ac77e01d11f35
> Reviewed-on: https://webrtc-review.googlesource.com/32001
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21225}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8566
Change-Id: I55611592c3171152cee97e64bff35a0d62cea510
Reviewed-on: https://webrtc-review.googlesource.com/33080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21283}
This CL removes the following GN variables: rtc_build_libyuv,
rtc_libyuv_dir (as requested in webrtc:7906).
It also removes some unneeded dependencies on //third_party/libyuv.
WebRTC targets were using public_deps to depend on //third_party/libyuv
and this created a build graph where targets that were depending on
//third_party/libyuv were not declaring the dependency to GN because
they were somehow getting it from another target that was exposing
//third_party/libyuv header files even if it wasn't directly depending
on it.
Bug: webrtc:8605, webrtc:7906
Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb
Reviewed-on: https://webrtc-review.googlesource.com/32201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21275}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This will require a manual roll to downstream projects, since
the //test:perf_test target was introduced.
This is a reland of 10a8e7a9b5
Original change's description:
> iOS: Save perf results under Documents/perf_result.json
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:7156
> Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5
> Reviewed-on: https://webrtc-review.googlesource.com/29202
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21244}
TBR=henrika@webrtc.org, phoglund@webrtc.org
No-Try: true
Bug: webrtc:7156
Change-Id: Iecdb108f605fd1c98acde4d50ab4f5a7b5f6bfaf
Reviewed-on: https://webrtc-review.googlesource.com/32761
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21252}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
Use std::fstream instead of rtc::File to write perf results.
On Android, when I use rtc::File, the results are not written for some reason.
Also rename the flag to '--chartjson_result_file'.
Bug: webrtc:8566
Change-Id: I32215e2233e18690c41050dfd35ac77e01d11f35
Reviewed-on: https://webrtc-review.googlesource.com/32001
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21225}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
instead of pair of pointer + size.
it removes hidden memcpy in RtpPacketReceived::Parse:
RtpPacketReceived keeps a reference to a CopyOnWriteBuffer. By
passing it the same CopyOnWriteBuffer that was created by
BaseChannel, one allocation and memcpy is avoided.
Bug: None
Change-Id: I5f89f478b380fc9aece3762d3a04f228d48598f5
Reviewed-on: https://webrtc-review.googlesource.com/23761
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21143}
The fuzzer will hammer on the UpdateSid and Generate methods of
ComfortNoiseDecoder.
The change also includes a fix to an issue in WebRtcSpl_FilterAR, which
was immediately found by running the fuzzer locally.
Bug: none
Change-Id: I5283427cb27844fb953e2caa35423ea873aca2ff
Reviewed-on: https://webrtc-review.googlesource.com/28100
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21134}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
The script that processes the RESULT lines doesn't support scientific notation [1],
so "1.234567e+06 units" is interpreted as "1.234567", "e+06 units".
Increase precision so that this is printed as 1234567 instead. I'll also submit a
CL so that the RESULT lines processor supports scientific notation.
[1] https://cs.chromium.org/chromium/build/scripts/slave/performance_log_processor.py?l=410
Bug: chromium:791501
Change-Id: If768d86b7ed07d92541ece6298eac8fe95880e35
Reviewed-on: https://webrtc-review.googlesource.com/29001
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21034}
Header files base/videosinkinterface.h and base/videosourceinterface.h
were not part of any target (because they cause 2 dependency cycles).
This CL uncomment them so GN can keep dependencies under control, the
2 dependency cycles will be removed as part of webrtc:6828.
Bug: webrtc:6828
Change-Id: I5c5580facc010ba619e105a9b8a572ac70169a01
Reviewed-on: https://webrtc-review.googlesource.com/27621
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20970}
This is a reland of 20f2133d5d
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This reverts commit 20f2133d5d.
Reason for revert: Breaks downstream project.
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org
Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.
This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
The random square generator produces unrealistically complex frames in
some situations, leading to frames > 250kb even at max QP. This leads to
unmanageably long transmission delays.
Bug: None
Change-Id: I8f5a33d52fb5efa03de97e529ad598b75511f679
Reviewed-on: https://webrtc-review.googlesource.com/23561
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20912}
This reverts commit b54bc06079.
Reason for revert: Downstream projects should be fixed now
Original change's description:
> Let PrintResultMeanAndError accept a string.
>
> Some downstream projects still use it. I'll update them and then revert
> this change.
>
> Bug: webrtc:8566
> Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
> Reviewed-on: https://webrtc-review.googlesource.com/25841
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20872}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
Change-Id: I5d2a59cbfb6a148fc6c621a69fa23397ba2c6991
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8566
Reviewed-on: https://webrtc-review.googlesource.com/25920
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20880}
Some downstream projects still use it. I'll update them and then revert
this change.
Bug: webrtc:8566
Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
Reviewed-on: https://webrtc-review.googlesource.com/25841
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20872}
It will be easier to make perf results output to JSON if the PrintResult*
functions receive doubles instead of strings.
I'll make follow-up CLs for PrintResultMeanAndError and PrintResultList.
Bug: webrtc:8566
Change-Id: I198e422a7bb8cd237c6364af98d2f67f0858452e
Reviewed-on: https://webrtc-review.googlesource.com/25300
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20848}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.
On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.
Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
This is temporarily needed while landing https://webrtc-review.googlesource.com/c/src/+/23820 and updating clients.
Bug: webrtc:4690
Change-Id: Ib0bd6a6a063a8a54c80b73853b2c042dfb02c44a
Reviewed-on: https://webrtc-review.googlesource.com/24501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20791}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=henrik.lundin@webrtc.org
Bug: None
Change-Id: I446549a385e020c68bafc83d5dd1aabd11d7ae18
Reviewed-on: https://webrtc-review.googlesource.com/23563
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20718}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
Don't include x11-specific code in test_renderer_generic when
use_x11 is set to false.
Bug: webrtc:8500
Change-Id: If64305e63484b985d90a9c9381bd391e34db3e26
Reviewed-on: https://webrtc-review.googlesource.com/21000
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20602}
Adds a constructor for DirectTransport that takes a pointer to an instance
of a class derived from FakeNetworkPipe. Said class can override Process()
and SendPacket(...) members thereby emulating any desired network behavior.
Bug: b/67487983
Change-Id: I829fd3506124db61587af19192a14fdf62b06ca5
Reviewed-on: https://webrtc-review.googlesource.com/14620
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20443}
When audio_processing_fuzzer runs with 'DCHECK_ALWAYS_ON', it crashes
when both AEC and AECM is enabled at the same time. This change
detects that case and fixes
https://clusterfuzz.com/v2/testcase-detail/6389429496446976.
It also removes an unnecessary safeguard that didn't allow fuzzing
with 8kHz input signals.
Bug: chromium:776358
Change-Id: I33c18a2a235e50ae410f7be24637872823e432eb
Reviewed-on: https://webrtc-review.googlesource.com/15320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20428}
NetEq has been fuzzed by neteq_rtp_fuzzer for some time. That fuzzer
hammers the RTP data, but leaves much of the other data alone. This
new fuzzer instead alters the encoded audio, packet arrival timing,
clock drift, and packet losses.
Bug: webrtc:8421
Change-Id: Ie25b77590a66a7451f32a73c6b5b570944244027
Reviewed-on: https://webrtc-review.googlesource.com/13860
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20368}
The struct containing the config for AEC3 is removed from
AudioProcessing::Config and is put in a new struct called
EchoCanceller3Config.
AEC3 should no longer be activated through
AudioProcessing::ApplyConfig. Instead an EchoCanceller3Factory
can be injected at AudioProcessing creation.
Bug: webrtc:8346
Change-Id: I27e3592e675eec3632a60c45d9e0d12514c2c567
Reviewed-on: https://webrtc-review.googlesource.com/11420
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20342}
Reading the fuzzer coverage report for audio_processing_fuzzer, I
noticed that AgcManagerDirect::AnalyzePreProcess was never
called. That turned out to be because GainControl was never
enabled. This change optionally calls 'Enable' on GainControl and 6
other public submodules of the APM.
Bug: webrtc:7820
Change-Id: Iae9da16f9f14fe7d3cd3318836d0d6e131a7ac39
Reviewed-on: https://webrtc-review.googlesource.com/12924
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20334}
This adds and extra param to SquareGenerator's constructor that sets the number of squares used. By default, it uses the same value that was previously hard-coded.
Bug: webrtc:8326
Change-Id: Ie7cff94e4a54fd5bb91f981930cad5e624e0e132
Reviewed-on: https://webrtc-review.googlesource.com/6020
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20223}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
This is a reland of 34cdd2d402
Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
>
> (Re-upload of https://codereview.webrtc.org/3020493002/)
>
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}
Bug: webrtc:4690, webrtc:7306
Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd
Reviewed-on: https://webrtc-review.googlesource.com/6300
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20137}
This reverts commit b7239a9dc8.
Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h.
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/5800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20107}
The refcount.h file doesn't depend on anything from
refcountedobject.h. The motivation of this change to make it possible
to add additional declarations to refcount.h, and include it from
refcountedobject.h.
Bug: webrtc:8270
Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
Reviewed-on: https://webrtc-review.googlesource.com/5760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20106}
On android, the flag to store the frame with the worst PSNR was called
'--test_artifacts_dir'.
I think test artifacts is a better name.
TBR=sprang@webrtc.org
Bug: chromium:745469
Change-Id: I358ea2985a1df2da12b81df173d74ac193556a49
Reviewed-on: https://webrtc-review.googlesource.com/4080
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20000}
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
- sqcif7 at 30 kbps: MediaCodec and libvpx.
- 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
BUG=webrtc:8219
Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This CL does not attempt to fix the style issues in the moved tb_ files, at least not yet. In general I've tried to avoid dependencies between the library and vie_auto_test: vie_auto_test depends on the library but not the other way around. I had to make some slight changes to achieve this. I had to remove some ViETest::Log statements in tb_interfaces.cc and I had to move knowledge of where to put output files to the library. I think it ended up being pretty clean in the end but let me know if I missed something. I tried to convert all paths I touched to src-relative paths, so look out if I missed something there.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1923 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose for the xvfb mode is to be able to run tests on the windowless environment on the Chromebot. Given the right input video, we can then write a relatively simple algorithm to analyze the screenshots and thereby conclude that video is playing.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/447004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1890 4adac7df-926f-26a2-2b94-8c16560cd09d
Restructured paths to new directory layout.
Stefan: common_video/*
Magnus: video_engine/*
Niklas: Android.mk
BUG=
TEST=jpeg_unittests on Debug+Release on Linux, Mac, Windows. Valgrind on Linux passes without warnings.
Review URL: https://webrtc-codereview.appspot.com/388007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1691 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed reproducable random behavior in packet_manipulator.h.
Test is now fully reproducable (runs on only one core) so much tighter limits are now set for the SSIM/PSNR values for the encoding/decoding (verified on all platforms)
BUG=
TEST=out/Debug/video_codecs_test_framework_integrationtests in Debug+Release on Linux, Mac, Windows and in Linux Valgrind.
Review URL: https://webrtc-codereview.appspot.com/381005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1649 4adac7df-926f-26a2-2b94-8c16560cd09d
The files are shorter (7 s) with one set provided for each sample rate.
Will be accompanied by the following set of files in the resource bundle:
far8_stereo.pcm
far16_stereo.pcm
far32_stereo.pcm
near8_stereo.pcm
near16_stereo.pcm
near32_stereo.pcm
BUG=114
TEST=audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/380003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
- Only showing text about browser needing WebRTC support if support not detected. Text is now contains more information and link to blog post.
- Removed the debug buttons.
- Clarifications and corrections in the readme file.
Review URL: https://webrtc-codereview.appspot.com/352015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1491 4adac7df-926f-26a2-2b94-8c16560cd09d
Performance verified on a few 32 kHz files.
BUG=
TEST=audioproc, audioproc_unittest
Updated output_data_float.pb
Changes in SWB tests (3, 6, 9 and 12) as
Running test 3 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 1363
Expected: test->max_output_average()
Which is: 1386
Running test 6 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 2070
Expected: test->max_output_average()
Which is: 2109
Running test 9 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 1314
Expected: test->max_output_average()
Which is: 1336
Running test 12 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 2049
Expected: test->max_output_average()
Which is: 2085
Review URL: https://webrtc-codereview.appspot.com/344013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1465 4adac7df-926f-26a2-2b94-8c16560cd09d
Example: .../webrtc_test.html?server=foo
This simplifies when one has to close and re-open the browser several times or use different servers and names, since it can be stored as bookmarks instead of changing it manually every time.
Review URL: http://webrtc-codereview.appspot.com/339006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1351 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.
BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/333025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.
TEST=audioproc_unittest + offline file tests.
output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.
TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
The recorded frame timestamps are used to modify the output video on a frame-per-frame so it can be compared with the reference video using PSNR. This code will make it possible to use vie_auto_test for full stack comparisons with network interference and similar interesting simulations.
There's some refactoring done in vie_comparison_test.cc to make it fit to the new test.
Compiled and executed in Debug+Release on Linux, Mac and Windows.
BUG=
TEST=vie_auto_test --automated --gtest_filter=ViEVideoVerificationTest.*
Review URL: http://webrtc-codereview.appspot.com/320002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1269 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.
Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.
Reworked all includes and GYP include paths to use full directory paths.
Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc
BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/311001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.
BUG=
TEST=audiproc, audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/292004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.
TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/279003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
Rewrote the codec test to render to file and do video comparisons.
Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.
Added video analysis to the test. This will make sure that the system output roughly the right thing.
Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.
Made sure no one passes in too large YUV videos into the autotest.
The standard test's output now gets captured for both the left and right windows.
Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/249001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
