Commit Graph

22 Commits

Author SHA1 Message Date
a8ae407a48 Add ability to cap the video jitter estimate to a max value.
Bug: webrtc:10572
Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27744}
2019-04-24 16:03:15 +00:00
6a489f22c7 Fully qualify googletest symbols.
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}
2019-04-09 17:18:20 +00:00
9b0b1e0063 Delete unused method VCMReceiveStatisticsCallback::OnReceiveRatesUpdated
Only interesting call deleted in cl
https://codereview.webrtc.org/2704183002.

Move call to QualitySample (used for bad call detection) to
OnRenderedFrame

Bug: webrtc:7408
Change-Id: I0e9ae2ed62fe19a282377cb840e38bd2aae8f3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128768
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27243}
2019-03-22 14:56:18 +00:00
493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api.
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}
2019-02-27 15:08:34 +00:00
77536a2b81 Rename EncodedImage::_length --> size_, and make private.
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}
2019-01-16 07:40:47 +00:00
61832dd018 Propagate spatial index to EncodedImage.
Set spatial index of assembled VP9 picture equal to spatial index of
its top spatial layer frame.

Bug: webrtc:10151
Change-Id: Iae40505864b14b01cc6787f8da99a9e3fe283956
Reviewed-on: https://webrtc-review.googlesource.com/c/115280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26075}
2018-12-20 15:21:22 +00:00
5546aef682 Vp9 flexible mode fixes
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
  and return several frames combined from FrameBuffer.

Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
2018-12-04 15:36:28 +00:00
648a7cefe1 Delete method EncodedFrame::GetBitstream, part 1
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}
2018-11-28 14:52:32 +00:00
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
6d2165036c Don't decode frames with an older timestamp than the last decoded timestamp.
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.

Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
2018-10-22 13:11:46 +00:00
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
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}
2018-08-21 09:15:51 +00:00
52f53d5419 Revert "Add Timestamp accessor methods to the EncodedImage class."
This reverts commit f34d467b03da4f20a1d036a20966fcad43d2433f.

Reason for revert: Seems to break downstream project.

Original change's description:
> Add Timestamp accessor methods to the EncodedImage class.
> 
> Bug: webrtc:9378
> Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
> Reviewed-on: https://webrtc-review.googlesource.com/82100
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23734}

TBR=brandtr@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org

Change-Id: I3aa0c0119426886bc583c918aae862eb7f4b6b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9378
Reviewed-on: https://webrtc-review.googlesource.com/85600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23739}
2018-06-26 11:52:45 +00:00
f34d467b03 Add Timestamp accessor methods to the EncodedImage class.
Bug: webrtc:9378
Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
Reviewed-on: https://webrtc-review.googlesource.com/82100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23734}
2018-06-26 09:40:18 +00:00
798b28279e Don't update internal state of the FrameBuffer2 when an undecodable frame is inserted.
Bug: chromium:844313
Change-Id: I034bcb47092815695084e37c81150bafbfbc6b9c
Reviewed-on: https://webrtc-review.googlesource.com/79944
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23577}
2018-06-12 09:26:09 +00:00
812ceafb5a Ensure render time is zero when playout delay is zero so that minimal latency in the render pipeline is ensured.
Bug: webrtc:9135
Change-Id: Id9ae8ec59536808ba8923c73dd46abfe3fa6fe79
Reviewed-on: https://webrtc-review.googlesource.com/75600
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23309}
2018-05-18 14:47:26 +00:00
0fa82a60e9 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId.
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.

Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
2018-03-19 15:13:11 +00:00
9771c5050d Clear the FrameBuffer if it's full and a keyframe is being inserted.
Bug: webrtc:7705, webrtc:8593, chromium:706599, chromium:807624
Change-Id: Ie4e3e217bc2930fe511f8b6ad3a36afed484ab5f
Reviewed-on: https://webrtc-review.googlesource.com/59321
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22304}
2018-03-06 09:11:11 +00:00
8c4fe16e4c Make Frame buffer not drop frames unnecessary
Now VCMTiming::MaxWaitingTime will not clip negative values. Thus frame
buffer will be able to distinguish between late frames and when waiting
cycle was simply interrupted by a new inserted frame right before the
waiting timer would expire.

Bug: webrtc:8917
Change-Id: I6b253f459fcb3a346064a103cc92ee332b074e1b
Reviewed-on: https://webrtc-review.googlesource.com/57741
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22210}
2018-02-27 16:31:29 +00:00
e7c891f953 Renamed FrameObject to EncodedFrame.
The plan is to:
 1. Move FrameObject to api/video.
 2. Rename FrameObject to EncodedFrame.
 3. Move EncodedFrame out of the video_coding namespace.

This is the 2nd CL.

Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
2018-02-22 16:12:48 +00:00
a194e58e79 Move sequence_number_utils.h to rtc_base/
Bug: webrtc:8440
Change-Id: I36e70da6ce70b95db7d3fce8b0013bff5c795bfc
Reviewed-on: https://webrtc-review.googlesource.com/14860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20429}
2017-10-25 12:33:57 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
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