This allows the script to be reliably run on the bots.
Bug: webrtc:8365
Change-Id: Ie4f216403c1f4fabe8db07f261fe06d1e64be0ae
Reviewed-on: https://webrtc-review.googlesource.com/42720
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21717}
I made two changes to video_quality_loopback_test to make it possible
to run it on swarming.
1. We need to know the path to frame_analyzer when we're generating
the build files, and it must be already present.
I made frame_analyzer a resource, so it's downloaded to a known path
before generating the build files.
2. The .zip files for apprtc and golang are downloaded and isolated.
The script now extracts them and installs AppRTC.
Passing task:
https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
Bug: chromium:755660
Change-Id: I34090897402421d5b7e29f21fbed354551197f92
Reviewed-on: https://webrtc-review.googlesource.com/40920
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21716}
This makes it possible to only inject 1 or 0 video codec factories when
consuming the API using the PeerConnectionFactory+Native header.
Bug: webrtc:7925
Change-Id: I671d8dcdbdf2198a31f3890ff6b416441bd32d48
Reviewed-on: https://webrtc-review.googlesource.com/42661
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21715}
The conversational_speech_generator tool now adjusts the level of
different speech segments.
Implementation:
The Turn and MultiEndCall::SpeakingTurn structs have an extra 'gain'
member. It's read and parsed in timing.cc and put in a Turn
struct. It's put in a SpeakingTurn struct in multiend_call.cc and read
and applied to the signal in simulator.cc
Bug: webrtc:7494
Change-Id: I9b82a896eb616c8b5ef14d41dfdfd085ef1d3fbb
Reviewed-on: https://webrtc-review.googlesource.com/26280
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21714}
This minor issue was found by the UBSAN fuzzer.
We have used the Godbolt compiler explorer to check that similar
changes produce identical compiled code.
Bug: chromium:803078
Change-Id: Ib3fa38c101d7bda53d8d39062cb2c0a55144305f
Reviewed-on: https://webrtc-review.googlesource.com/42580
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21713}
AudioFrameOperations was moved from modules/utility to audio/utility
over 1 year ago. This CL finally removes the
modules/utility/include/audio_frame_operations.h header and closes the
bug.
TBR=perkj@webrtc.org
Bug: webrtc:6548
Change-Id: I36230eadef2f8edb81008562fdc9208cf29845a4
Reviewed-on: https://webrtc-review.googlesource.com/36140
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21712}
WebRTC is still targeting C++11, this C++14 only feature sneaked in
because the Android toolchain used to build WebRTC on the trybots
uses C++14 features.
Bug: None
Change-Id: I095fb76134dff729c72b7660cdb3d6abc4de2e0c
Reviewed-on: https://webrtc-review.googlesource.com/42501
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21711}
This reverts commit ce804ddd72f768781654a996b0f6a9551d5f2efa.
Reason for revert: chromium:800732 has been fixed.
Original change's description:
> Remove gradle from DEPS.
>
> The gradle git repo seems to be broken, so remove it from
> DEPS until it is fixed.
>
> Bug: webrtc:8724
> Change-Id: I718d3faadf9c636df8e840b0f8d32c52a73d7da4
> Reviewed-on: https://webrtc-review.googlesource.com/38600
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21549}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8724
Change-Id: Iad03a9e00320831833c589aa52643225df4e32a2
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/42480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21707}
Various places are using "memset(ptr, 0, size)" which might get optimized
away by the compiler if "ptr" is not used afterwards. The new functions
can be used to securely clear memory instead.
Bug: None
Change-Id: I067a51d17ff84d95dc4934d46a24027fbcb4825d
Reviewed-on: https://webrtc-review.googlesource.com/35500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21706}
Use EncodeBatch method in unittest. (Same as in production code.)
Bug: webrtc:8111
Change-Id: Ia194f5138f244da7f348821277f6c712a3ffab0d
Reviewed-on: https://webrtc-review.googlesource.com/34560
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21696}
This allows a user to only injecting the decoder or encoder factory.
This behavior also matches how it is implemented for Android.
Bug: webrtc:7925
Change-Id: I3dfca6ea2eaeea437b5b78da2373bd6f7cedc8fa
Reviewed-on: https://webrtc-review.googlesource.com/40860
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21695}
It is part of our api.
With the intention to later delete the inclusion of mediachannel.h from
api/peerconnectioninterface.h, and eliminate circular dependencies.
Bug: webrtc:7504
Change-Id: If44efd14d85675530e457760a1c4a1d338f931b7
Reviewed-on: https://webrtc-review.googlesource.com/41281
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@{#21694}
No stats is logged in this format any longer.
Bug: none
Change-Id: I5f91e93636b6d03ebd91c3b2518857275fb94de7
Reviewed-on: https://webrtc-review.googlesource.com/40700
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21690}
This is a reland of 18c4261339dc76b220e7c805e36b4ea6f3dd161d
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}
Current checks do not allow first frame after fallback to be sent for encode
because of the native checks. However, the rest of the frames are sent and
encoded correctly.
Fallback encoder, although SupportsNativeHandle() is false, can call ToI420()
which converts from native handles, and accordingly can return error or
success. This higher level check seems unnecessary.
Bug: webrtc:8021
Change-Id: I69780cc25eb1e06317ff213e9b80288064e9f1e3
Reviewed-on: https://webrtc-review.googlesource.com/40441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21685}
It's reference counted, yet we aren't taking a reference to it for some
reason. This could be causing it to be dereferenced after deletion in
some cases in chromium.
Bug: chromium:798251
Change-Id: I0b91451e38ed611d2ea8a477f1e7db482a790f79
Reviewed-on: https://webrtc-review.googlesource.com/37283
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21684}
This also changes RtpReceiver and RemoteAudioSource to have two-step
initialization, since in Unified Plan RtpReceivers are created much
earlier than in Plan B.
Bug: webrtc:7600
Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4
Reviewed-on: https://webrtc-review.googlesource.com/39382
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21681}
This used to be used by the linter, but now it's actually using
build/android/AndroidManifest.xml to determine the minimum SDK
level we support.
Remove this file to avoid confusion.
Bug: None
Change-Id: I2ea60854d52276877ae9d51a9c1f379cfaa3ed5d
Reviewed-on: https://webrtc-review.googlesource.com/40661
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21677}
GetPacingFactor exposed internal details that should not be relied upon.
In a later CL theese won't be available any more, this CL is in
preparation for that change.
The only usage was in video send stream tests. To keep the tests
working, they now access the internal video send stream directly. The
test code retrieves an optional that indicates whether the send stream
has overridden the pacing factor. This means the implementation
dependency between video send stream and video send stream tests is
increased. This is an improvement compared to depending on the paced
sender implementation.
Bug: webrtc:8415
Change-Id: Id357553692b3ff3283fa3b64da1b1ebb3c97f04d
Reviewed-on: https://webrtc-review.googlesource.com/39265
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21675}
These features are supported via desugar, but we started getting lint errors:
"Static interface method requires API level 24 (current min is 16)"
60990c11ce
Bug: webrtc:8084
Change-Id: Iba8b5825a52bd57da7ee4ab52cde2d7bd9ebce30
Reviewed-on: https://webrtc-review.googlesource.com/40421
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21672}
And wire it up to methods on RTCConfiguration, via MediaConfig::Video.
Bug: webrtc:8504
Change-Id: I30805ee20c11d1d2fe552eb81f16d514db0ba4a8
Reviewed-on: https://webrtc-review.googlesource.com/39786
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@{#21670}
