To eliminate circular dependencies, we need to eliminate the include
of media/base/mediachannel.h from api/peerconnectioninterface.h.
MediaConfig is one of the types the PeerConnection api depends on,
since it's part of PeerConnectionInterface::RTCConfiguration. It's
formally a public member, but the intention is that applications should use
accessor mehtods on RTCConfiguration and never access the contents of
MediaConfig directly.
Bug: webrtc:7504
Change-Id: Idfab6f69132d6b90d1628fa4543a393e22db79ac
Reviewed-on: https://webrtc-review.googlesource.com/41260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21731}
In crrev.com/531028, the JNI generator starts to add heap profiler
events to JNI generated functions.
This will cause a ~80KiB regression and at the moment it is breaking
the Chromium Roll into WebRTC.
This CL defines a void macro to re-enable the Chromium Roll avoiding
the size regression.
Bug: chromium:801260
Change-Id: I9543299199c4e14b6b9b235c5cb98c0d53cf29ea
Reviewed-on: https://webrtc-review.googlesource.com/43021
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21730}
Try to use frame timestamps first if they look reasonable, otherwise
use realtime clock.
Also, lower limit from 90% of target to 85%.
Bug: webrtc:4172, chromium:802290
Change-Id: Iad489be7c7cf637345be4795e5089936ab9fab07
Reviewed-on: https://webrtc-review.googlesource.com/41041
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21729}
This CL increases the speech of the initial alignment in AEC3 by
loosening the requirements on the accuracy of the initial estimates.
Bug: webrtc:8784, chromium:804270
Change-Id: I86e2d97830843524090a1cf877965739f66dc058
Reviewed-on: https://webrtc-review.googlesource.com/40660
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21728}
This reverts commit 1175ecd09599194caa05ecafdc4503e940809fc9.
Reason for revert: Breaks the bots.
Original change's description:
> Make it possible to run video_quality_loopback_test in swarming.
>
> 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}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
Change-Id: Id25d26adc547ff6f9ab178601e37527459c8b5ef
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/42800
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21723}
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}
The custom template seems to be broken. The necessary flag
"use_unprocessed_jars" has been added to dist_jar template and the
custom template is not needed anymore.
Bug: None
Change-Id: I0ca7a91ee47c8de659bcaffa5661bff55af50375
Reviewed-on: https://webrtc-review.googlesource.com/42680
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21721}
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 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}
