Commit graph

78 commits

Author SHA1 Message Date
Sergey Silkin
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
Ilya Nikolaevskiy
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
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
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}
2018-11-28 18:25:07 +00:00
Ilya Nikolaevskiy
e6a2d94eca Clear FrameBuffer if there were no frames received for 10 minutes
This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.

Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}
2018-11-07 15:09:11 +00:00
Niels Möller
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
philipel
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
“Michael
f9fc171568 Add rtt_mult_experiment to evaluate video robustness vs. latency
Bug: webrtc:9670
Change-Id: Idb4ca130bfa652b2d0bddb5bee9ed8e34c97150a
Reviewed-on: https://webrtc-review.googlesource.com/96060
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24457}
2018-08-27 15:51:52 +00:00
Niels Möller
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
Björn Terelius
52f53d5419 Revert "Add Timestamp accessor methods to the EncodedImage class."
This reverts commit f34d467b03.

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
Niels Möller
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
Danil Chapovalov
0040b66ad3 Replace rtc::Optional with absl::optional
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':

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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:24:48 +00:00
philipel
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
Rasmus Brandt
45a57fda24 Remove unused include from FrameBuffer2.
Bug: None
Change-Id: I766b430beb4f5ba35519931fbff19261a462f2c2
Reviewed-on: https://webrtc-review.googlesource.com/81184
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23517}
2018-06-05 11:33:20 +00:00
Stefan Holmer
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
Niels Möller
be682d47ac Fix chromium warnings for SdpVideoFormat.
Bug: webrtc:163
Change-Id: I29ad3c00116692f047456df7721ba636bbb2ca89
Reviewed-on: https://webrtc-review.googlesource.com/64723
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22618}
2018-03-27 08:11:21 +00:00
philipel
9718711dee VideoStreamDecoderImpl implementation, part 1.
In this CL the OnFrame function is implemented.

Bug: webrtc:8909
Change-Id: I68488a033e86eadd0b16d091faad14e9cda7cc36
Reviewed-on: https://webrtc-review.googlesource.com/64121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22583}
2018-03-23 13:58:55 +00:00
philipel
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
philipel
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
philipel
0a9f6de446 Removed VCMTiming from RtpVideoStreamReceiver.
Bug: webrtc:8909
Change-Id: Ib42e4cc4c2252f04ea5f7d440352546d36d2899f
Reviewed-on: https://webrtc-review.googlesource.com/58740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22224}
2018-02-28 11:31:51 +00:00
Ilya Nikolaevskiy
7eef00719b Fix typo in FrameBuffer
Bug: none
Change-Id: Ifc9a531da9460b7cac4aa71fb468c0881a663e94
Reviewed-on: https://webrtc-review.googlesource.com/58641
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22223}
2018-02-28 11:29:30 +00:00
Ilya Nikolaevskiy
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
philipel
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
philipel
1610f94ee3 Don't cast picture ids (of type int64_t) to int.
Also cleaned up a bit in RtpFrameReferenceFinder.

Bug: chromium:762556
Change-Id: Ib08d2e7ce4b146b359ce9ba823f3aa15776c71bc
Reviewed-on: https://webrtc-review.googlesource.com/32301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21282}
2017-12-14 14:22:13 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
philipel
707f278299 Add RTT to playout delay behind WebRTC-AddRttToPlayoutDelay field trial.
Bug: webrtc:8010
Change-Id: I78d2b5053521186b9bcc27eba264325b6f934a78
Reviewed-on: https://webrtc-review.googlesource.com/4666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20079}
2017-10-02 13:28:30 +00:00
philipel
e21be1db4c Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reason for revert:
Fixes has landed.

Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
2017-09-25 13:37:12 +00:00
Mirko Bonadei
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
Mirko Bonadei
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
Renamed from webrtc/modules/video_coding/frame_buffer2.cc (Browse further)