Commit Graph

22 Commits

Author SHA1 Message Date
f5033ad201 In (new) estimator of encoder cpu load, count max per input frame.
Bug: webrtc:9619
Change-Id: Ifc874fa3bd069e48a10fec57b673546aafd070e3
Reviewed-on: https://webrtc-review.googlesource.com/94143
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24293}
2018-08-15 15:16:47 +00:00
58d2a5e976 Tolerate out of order samples to SendProcessingUsage2::FrameSent.
Bug: chromium:842613
Change-Id: I57e4df75dcfdfb9bf42819f31d2186e875a90a3a
Reviewed-on: https://webrtc-review.googlesource.com/92880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24224}
2018-08-08 09:58:42 +00:00
213618e37e New api function CreateVideoStreamEncoder.
Bug: webrtc:8830
Change-Id: I01de86f601e48f76e6b41b4182ce006d397a190c
Reviewed-on: https://webrtc-review.googlesource.com/78260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24079}
2018-07-24 09:14:26 +00:00
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
665174fdbb Reformat the WebRTC code base
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}
2018-06-19 14:00:39 +00:00
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
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: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00
4db138e889 Reland "Move creating encoder to VideoStreamEncoder."
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80

Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> 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@{#22905}

TBR=magjed@webrtc.org,kwiberg@webrtc.org

Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
2018-04-19 08:48:58 +00:00
0d650b44ef Revert "Move creating encoder to VideoStreamEncoder."
This reverts commit fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80.

Reason for revert: Appears to break Chromium, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Linux%20Tester/43756, where remoting_unittests failed.

Original change's description:
> Move creating encoder to VideoStreamEncoder.
> 
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
> 
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> 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@{#22905}

TBR=magjed@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: I47ee3ac42e62472d825a08c98e28f9ae53ec9fff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/70600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22914}
2018-04-18 07:17:16 +00:00
fb82fcc7f9 Move creating encoder to VideoStreamEncoder.
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.

Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
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@{#22905}
2018-04-17 15:04:33 +00:00
d1f7eb6e83 Postpone setting of CpuOveruseOptions.
This will enable changing thresholds when switching between hardware
and software encoders. It is also a partial revert of
https://webrtc-review.googlesource.com/33340: construction of the
OveruseFrameDetector is still in VideoSendStream, but configuration is
moved back to VideoStreamEncoder.

Longer term, information about HW vs SW, or generally, about resources
consumed by the encoder, should be passed in the per-frame callbacks
to OveruseFrameDetector, and then the CpuOveruseOptions could move
back to construction time.

Bug: webrtc:8504, webrtc:8830
Change-Id: I44577519d4e05356730cac9bd9ae3c74bfc17ed7
Reviewed-on: https://webrtc-review.googlesource.com/65163
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22761}
2018-04-06 08:34:32 +00:00
73f29cbcc1 Move creation of OveruseFrameDetector to VideoSendStream.
Intended to make it easier to wire up cpu-adaptation experiments.
To setup the circular references between OveruseFrameDetector and
VideoStreamEncoder, let the AdaptationObserverInterface pointer be
an argument to StartCheckForOveruse.

Bug: webrtc:8504
Change-Id: Ifcf7655ec65e637819d77f507552cb22a6aa5f0f
Reviewed-on: https://webrtc-review.googlesource.com/33340
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22006}
2018-02-13 15:45:03 +00:00
5e4833cc90 Add missing stdio.h header in files using scanf/sscanf function.
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.

Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
2018-01-26 13:15:52 +00:00
8e07c134ab Optional: Use nullopt and implicit construction in /video
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: Ie622c215e06956d8d5629733c76f531b7af45012
Reviewed-on: https://webrtc-review.googlesource.com/23568
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21535}
2018-01-09 15:14:10 +00:00
83dbeacb1a Add alternative load estimator to OverUseFrameDetector.
The new estimator uses the timestamps attached to EncodedImage, and is
taken from the reverted cl
https://webrtc-review.googlesource.com/c/src/+/23720.

Bug: webrtc:8504
Change-Id: I273bbe3eb6ea2ab9628c9615b803a379061ad44a
Reviewed-on: https://webrtc-review.googlesource.com/31380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21289}
2017-12-15 10:10:06 +00:00
6b642f730c Delete EncodedFrameObserver::OnEncodeTiming.
This callback was used only by the PrintSamplesToFile feature of
video_quality_test, which looks like it has been broken for some time
(due to mixup of capture time and ntp time).

Bug: webrtc:8504
Change-Id: I7d2b55405caeffda582ae0d6fb0e7dfdfce4c5a9
Reviewed-on: https://webrtc-review.googlesource.com/31420
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21211}
2017-12-11 16:41:46 +00:00
e08cf3a615 Move FrameTiming list from OveruseDetector to SendProcessingUsage.
Bug: webrtc:8504
Change-Id: Ifd8feb1634f1061f841493eae83c9eae410d4c20
Reviewed-on: https://webrtc-review.googlesource.com/30780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21153}
2017-12-08 07:57:49 +00:00
904f8698d3 Define interface OveruseFrameDetector::ProcessingUsage.
Allows move of SendProcessingUsage and OverdoseInjector to the
anonymous namespace in the implementationfile.

Bug: webrtc:8504
Change-Id: I84ddfc0b031a1fd31cc0f35216ba96c8fb2efa8e
Reviewed-on: https://webrtc-review.googlesource.com/30682
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21138}
2017-12-07 13:22:19 +00:00
7dc26b7b32 Revert "Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage."
This reverts commit eee7cedf6901ad5616dbf0bf09f35010207f823d.

Reason for revert: Intend to rework with a flag to select between old or new estimator, to be wired up as an origin-trial experiment. And old estimator deleted at a later point.

Original change's description:
> Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage.
> 
> Bug: webrtc:8504
> Change-Id: I3f99c3c1e528f59b45724921a91f65b485f5b820
> Reviewed-on: https://webrtc-review.googlesource.com/23720
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20979}

TBR=nisse@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8504
Change-Id: Ieec4624e1d6dd8472b7e89c7bd19f425d9b54533
Reviewed-on: https://webrtc-review.googlesource.com/30180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21115}
2017-12-06 10:32:12 +00:00
eee7cedf69 Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage.
Bug: webrtc:8504
Change-Id: I3f99c3c1e528f59b45724921a91f65b485f5b820
Reviewed-on: https://webrtc-review.googlesource.com/23720
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20979}
2017-12-01 15:13:26 +00:00
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
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