This reverts commit 87bed4793f.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Reland "Preserve min and max playout delay from RTP header extension"
>
> This reverts commit f31cc08ba0.
>
> Reason for revert: Reland with fixes
>
> Original change's description:
> > Revert "Preserve min and max playout delay from RTP header extension"
> >
> > This reverts commit 85ba9972c4.
> >
> > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
> >
> > Original change's description:
> > > Preserve min and max playout delay from RTP header extension
> > >
> > > Audio and video synchronization can sometimes override the minimum
> > > and maximum playout delay that is set through the RTP header
> > > extension. This CL makes sure that the playout delay always is
> > > within the limits set by the RTP header extension.
> > >
> > > Bug: webrtc:10886
> > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28980}
> >
> > TBR=stefan@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:10886
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28984}
>
> TBR=stefan@webrtc.org,kron@webrtc.org
>
> Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28985}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28986}
This reverts commit f31cc08ba0.
Reason for revert: Reland with fixes
Original change's description:
> Revert "Preserve min and max playout delay from RTP header extension"
>
> This reverts commit 85ba9972c4.
>
> Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
>
> Original change's description:
> > Preserve min and max playout delay from RTP header extension
> >
> > Audio and video synchronization can sometimes override the minimum
> > and maximum playout delay that is set through the RTP header
> > extension. This CL makes sure that the playout delay always is
> > within the limits set by the RTP header extension.
> >
> > Bug: webrtc:10886
> > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28980}
>
> TBR=stefan@webrtc.org,kron@webrtc.org
>
> Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10886
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28984}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28985}
This reverts commit 85ba9972c4.
Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream.
Original change's description:
> Preserve min and max playout delay from RTP header extension
>
> Audio and video synchronization can sometimes override the minimum
> and maximum playout delay that is set through the RTP header
> extension. This CL makes sure that the playout delay always is
> within the limits set by the RTP header extension.
>
> Bug: webrtc:10886
> Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28980}
TBR=stefan@webrtc.org,kron@webrtc.org
Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28984}
Audio and video synchronization can sometimes override the minimum
and maximum playout delay that is set through the RTP header
extension. This CL makes sure that the playout delay always is
within the limits set by the RTP header extension.
Bug: webrtc:10886
Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28980}
This change re-enables a previously flaky unit tests for iOS. It seems to have the same root cause as webrtc:10827 and which was fixed by: https://webrtc-review.googlesource.com/c/src/+/149171
Bug: webrtc:10872, webrtc:10827
Change-Id: I71b2581cf8c75e0dd6a39b77e6cf34c121ff22f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149802
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28893}
This partially reverts these 2 CLs:
1) Reland "Copy video frames metadata between encoded and plain frames in one place"
https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9
2) Don't copy video frame metadata in each encoder/decoder
https://webrtc.googlesource.com/src/+/ab62b2ee51e622be6d0aade15e87e927fa60e6f2
The problem with them were that ColorSpace was made to always be copied from the
EncodedImage in the GenericDecoder, which overwrote ColorSpace information from
the decoder.
If decoder applied color space transition or bitstream color space information
was different from the WebRTC signaled one, the incorrect color space data were
passed to the renderer.
This CL removes introduced change regarding color space data: GenericDecoder
doesn't copy or store it and software decoders are restored to copy it.
Relevant tests are also removed.
Bug: chromium:982486
Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145323
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28556}
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.
The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
This reverts commit 4fb12b0cae.
Reason for revert: Breaks some asan chromium bots
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27828}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10460
Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27895}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27828}
This reverts commit c9a2c5e93a.
Reason for revert: Breaks downstream test
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27756}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
Change-Id: I34cc563ec6383735c2a76a6f45a72a7726b74421
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134204
Reviewed-by: Artem Titarenko <artit@google.com>
Commit-Queue: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27765}
Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27756}
This reverts commit 00d0a0a1a9.
Reason for revert: Breaks downstream tests
Original change's description:
> Copy video frames metadata between encoded and plain frames in one place
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Bug: webrtc:10460
> Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27719}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org
Change-Id: I8960a6cc15e552925129ba0037f197ff3fd93c25
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134100
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27737}
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Bug: webrtc:10460
Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27719}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
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}
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}
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}
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}
Reduce synchronization in the class significantly and not hold a lock
while calling out to external implementations.
* Rewrite tests to use a real ProcessThread.
* Update some code to use C++ 11 constructs & library features.
Bug: webrtc:9064
Change-Id: I240a819efb6ef8197da3f2edf7acf068d2a27e8b
Reviewed-on: https://webrtc-review.googlesource.com/64521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22649}
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/video/video_receive_stream_unittest.cc (Browse further)
