1941 Commits

Author SHA1 Message Date
Erik Språng
f141470d38 Store qp limits for ScreenshareLayers only once
Bug: webrtc:9745
Change-Id: Ie38b9d4991100657c1dc54660b39b80d86cc64fa
Reviewed-on: https://webrtc-review.googlesource.com/99940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24716}
2018-09-13 08:53:10 +00:00
Jonas Olsson
941a07cca3 Remove all remaining non-test uses of std::stringstream.
Bug: webrtc:8982
Change-Id: I635a8545c46dc8c89663d64af351e22e65cbcb33
Reviewed-on: https://webrtc-review.googlesource.com/98880
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24715}
2018-09-13 08:52:05 +00:00
Niels Möller
72bc8d6df6 Make the rtp timestamp member of EncodedImage private
A followup to https://webrtc-review.googlesource.com/c/src/+/82160,
which added accessor methods.

Bug: webrtc:9378
Change-Id: Id3cff46cde3a5a3fb6d6edd4e8dac26193e6481c
Reviewed-on: https://webrtc-review.googlesource.com/95103
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24705}
2018-09-12 13:44:36 +00:00
Minyue Li
1a80018a3c Avoid wrong parsing of padding length and its use in NetEq simulation.
Bug: b/113648474, webrtc:9730
Change-Id: Ieff7ab8697f5c8742548897a9b452a20b0bd2e7c
Reviewed-on: https://webrtc-review.googlesource.com/98461
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24703}
2018-09-12 11:23:03 +00:00
Danil Chapovalov
fd5fbd0b58 Cleanup RtpPacketizerH264 constructor
Merge SetPayloadData into constructor.
Add TODO to support first_packet_reduction_len

Bug: webrtc:9680
Change-Id: I65e771848e0ffe8968cd084840e77afc0152caeb
Reviewed-on: https://webrtc-review.googlesource.com/99505
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24702}
2018-09-12 11:11:18 +00:00
Jose Antonio Olivera Ortega
289e980708 Remove unused var in device info bits from video capture module for Linux
Bug: None
Change-Id: Icea40fe58e7f65cd1eb311c456ce3cdc802f88a8
Reviewed-on: https://webrtc-review.googlesource.com/97421
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24700}
2018-09-12 10:36:33 +00:00
Kári Tristan Helgason
640106e1ce Use different thresholds for ARM and x86 in libvpx tests
and audio processing tests.

Bug: webrtc:8757
Change-Id: Ic748fa624ac84af4556cb4b51718106a10fbb787
Reviewed-on: https://webrtc-review.googlesource.com/98540
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24698}
2018-09-12 08:18:33 +00:00
Johnny Lee
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
Danil Chapovalov
c7fff58d1e Allow nullptr retransmition rate limiter
as iniditcation retransmission shouldn't be limited because of rate.

Bug: None
Change-Id: I579261749515260b972631779dadc6349dfcab46
Reviewed-on: https://webrtc-review.googlesource.com/99541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24690}
2018-09-11 14:50:54 +00:00
Per Kjellander
e9da5f27a4 Reland "Decrease complexity of RtpPacketHistory::GetBestFittingPacket.""
This reverts commit 49b2c3c4c43359bc86d8510d29d117f3d7a621a3.

Original CL description:
Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call

patch set 1 contains the initial submit from https://webrtc-review.googlesource.com/c/src/+/98882
new patch sets contains the modification.

The problem with the initial submit was the assumption that packets are removed
from history in the same order as they are added which is not always true.

Bug: webrtc:9731
Change-Id: Ic2c8905a0f47287fc46e53f41a019a4c69c3dd8e
Reviewed-on: https://webrtc-review.googlesource.com/99460
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24687}
2018-09-11 13:54:30 +00:00
Sergey Silkin
beba1b2766 Fix target frame rate of spatial layer.
Set target frame rate of spatial layer equal to minimum of two: maximum
frame rate of layer (SpatialLayer::maxFramerate) and maximum frame rate
of codec (VideoCodec::maxFramerate).

Bug: webrtc:9740, webrtc:9739, chromium:882358
Change-Id: I34f36e7fd2889f0417474347abab5327fa2d9d7c
Reviewed-on: https://webrtc-review.googlesource.com/99501
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24686}
2018-09-11 13:50:58 +00:00
Erik Språng
375d35e01b Make ilnik@ owner in video/ and modules/video_coding/
Bug: None
Change-Id: I509b95fb70227d21d288716a886d5476f5242708
Reviewed-on: https://webrtc-review.googlesource.com/99581
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24685}
2018-09-11 12:39:39 +00:00
Gustaf Ullberg
ddb82a6b5f AEC3: Fix filter output transition when input and output is the same array
This CL fixes a bug in the filter output transition when the 'from' input
points to the same array as the output. It also includes a slight
improvement to the transition by starting one sample earlier than
previously.

Bug: webrtc:9741,chromium:882789
Change-Id: Ifd5f16c1ac88a74d93499e7f4b4c0e5cb3e4976f
Reviewed-on: https://webrtc-review.googlesource.com/99540
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24683}
2018-09-11 11:59:12 +00:00
Mirko Bonadei
7255fef258 Fix no_global_constructors in remote_bitrate_estimator.
Bug: webrtc:9693
Change-Id: Ibd8f0e89a7b37ad26d4cb3e73c395f77ed988ac9
Reviewed-on: https://webrtc-review.googlesource.com/98584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24682}
2018-09-11 11:44:32 +00:00
Gustaf Ullberg
51ccdbeb0c AEC3: Bugfix in filter output transition
Bug: webrtc:9741,chromium:882789
Change-Id: Id83f31dfa2cfaf06f41673ac997becf1e399eeea
Reviewed-on: https://webrtc-review.googlesource.com/99502
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24677}
2018-09-11 10:30:08 +00:00
Sami Kalliomäki
3d50a31aad Remove redundant initializers from WebRTC Java code.
Removes redundant field initializers such as null, 0 and false.

Bug: webrtc:9742
Change-Id: I1e54f6c6000885cf95f7af8e2701875a78445497
Reviewed-on: https://webrtc-review.googlesource.com/99481
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24676}
2018-09-11 09:58:10 +00:00
Ivo Creusen
f81b0f11a6 Move code for setting field trials from NetEqTestFactory to the main function in neteq_rtpplay.
It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay.

Bug: webrtc:9667
Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae
Reviewed-on: https://webrtc-review.googlesource.com/98541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24673}
2018-09-11 09:27:11 +00:00
Niels Möller
b0fc70303d Delete deprecated simulcast and spatial index members
These attributes were moved from CodecSpecificInfo to EncodedImage in
cl https://webrtc-review.googlesource.com/c/src/+/96780. This followup
deletes the old member variables, which were left temporarily to
transition downstream code.

Bug: webrtc:9378
Change-Id: I1b38ce404a005aec9d48916b73233cfbd7523cfe
Reviewed-on: https://webrtc-review.googlesource.com/97021
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24670}
2018-09-11 08:11:16 +00:00
Sergey Silkin
96f2c97161 Reland "Frame rate controller per spatial layer."
This is a reland of ae9e188e67a489db597224e3cfcfdee04edf0cba

Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}

TBR=sprang@webrtc.org

Bug: webrtc:9682
Change-Id: Idcce315890c79301da532f9ba4997e9606f73fb0
Reviewed-on: https://webrtc-review.googlesource.com/99340
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24669}
2018-09-11 07:08:36 +00:00
Mirko Bonadei
8b0aed1dd6 Fix no_global_constructors/no_exit_time_destructors in Neteq.
Bug: webrtc:9693
Change-Id: I0135e934c638ec391546928ba1e623d137b27b75
Reviewed-on: https://webrtc-review.googlesource.com/98600
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24668}
2018-09-11 06:39:14 +00:00
Mirko Bonadei
64ef4f7c95 Fix no_global_constructors in audio_processing/agc2/rnn_vad.
This is a rework of [1] following kwiberg@'s advice.

[1] - https://webrtc-review.googlesource.com/c/src/+/98583

Bug: webrtc:9693
Change-Id: I8d4fac8d7593c28d4ad2a973637f965f2cd51e99
Reviewed-on: https://webrtc-review.googlesource.com/98881
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24667}
2018-09-11 06:23:56 +00:00
Mirko Bonadei
04255172b6 Remove double declaration of 2 conversion functions.
The declaration in common_types.h is probably a left-over from a
previous cleanup.

Bug: None
Change-Id: I3ee1bad2494ede0022c6aa8fdd106035471d50e2
Reviewed-on: https://webrtc-review.googlesource.com/99220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24666}
2018-09-11 06:00:05 +00:00
Per Kjellander
49b2c3c4c4 Revert "Decrease complexity of RtpPacketHistory::GetBestFittingPacket."
This reverts commit 54caa4b68a0acb81c7f6ef60ffec45b473a7e1a2.

Reason for revert: Crashes on some perf tests.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc.perf%2FLinux_Trusty%2F7170%2F%2B%2Frecipes%2Fsteps%2Fwebrtc_perf_tests%2F0%2Fstdout



Original change's description:
> Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
> Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.
> 
> Bug: webrtc:9731
> Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
> Reviewed-on: https://webrtc-review.googlesource.com/98882
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24662}

TBR=danilchap@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: Id183cd31a67117e9614d163e4388131fd88de07d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9731
Reviewed-on: https://webrtc-review.googlesource.com/99440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24665}
2018-09-11 05:35:09 +00:00
Per Kjellander
54caa4b68a Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.

Bug: webrtc:9731
Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
Reviewed-on: https://webrtc-review.googlesource.com/98882
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24662}
2018-09-10 19:09:29 +00:00
Sebastian Jansson
57606328f6 Adds initial data window field trial to GoogCC.
Bug: webrtc:9718
Change-Id: Ia5a77a09d7ba82b545e9ab12036f717765fdf3b4
Reviewed-on: https://webrtc-review.googlesource.com/97740
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24659}
2018-09-10 16:55:37 +00:00
Henrik Lundin
9be7745509 NetEq tools: Fixing an issue with measuring the simulation time
The NetEqTest class was recently refactored. In the process, the
functionality for measuring the simulation time suffered a bug. This
CL fixes it.

Bug: webrtc:9667
Change-Id: I139e697ede21584ef77ae23cfa8e77f6dac65b51
Reviewed-on: https://webrtc-review.googlesource.com/98982
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24658}
2018-09-10 16:16:22 +00:00
Alex Loiko
d934244feb Added flags for the adaptive analog AGC in audioproc_f.
Added back the 'agc2 level estimation' flag. Also added a flag for
moving the level measurement before AEC and NS. This is to run offline
experiments with audioproc_f.


Bug: webrtc:7494
Change-Id: I3e3ffceede7166b754130be2b707b620ba527e9f
Reviewed-on: https://webrtc-review.googlesource.com/97442
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24657}
2018-09-10 14:16:46 +00:00
Per Åhgren
b2d7116733 AEC3: Correction of the suppressor behavior at delay changes
This CL adjusts the behavior of the AEC3 echo suppressor behavior
initially in the call, and when there has been delay changes. The
results is that short echo blips/bursts present in some such cases
no longer occur.

In particular this CL:
-Ensures that the suppressor back-off under stationary render
conditions does not occur until the linear filter has had the
ability to converge.
-Ensures that a previously converged filter behavior detection
is not sticky for stable and linear echo paths, which in turn
prevents echo leakage due to the more liberal echo suppressor
behavior applied on such platforms.
-Removes a bug that caused a random and jittery behavior for
the usage of the linear filter output initially in the calls
and after echo path changes

Bug: webrtc:9737, chromium:882396
Change-Id: Id2b46e366dc58ab8137f19ed59a2034c89ca3087
Reviewed-on: https://webrtc-review.googlesource.com/99063
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24656}
2018-09-10 13:05:14 +00:00
Danil Chapovalov
e4f8b38091 Allow different header extensions in 1st packet of a video frame
no behavior changes expected.
Different exension for the 1st packet will be added in a follow-up

Bug: webrtc:9680
Change-Id: I8c853b2710d58df579aeb4b029b42210310423cc
Reviewed-on: https://webrtc-review.googlesource.com/98843
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24655}
2018-09-10 12:53:10 +00:00
Danil Chapovalov
067818fe85 Move RtcpTransceiver deletion of copy and assign methods to public section
Bug: chromium:881453
Change-Id: Iff5c522b983af018c1308649887a1121519c73ea
Reviewed-on: https://webrtc-review.googlesource.com/98981
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24652}
2018-09-10 12:23:19 +00:00
Alex Loiko
623472219f Store RuntimeSetting in Aec Dumps.
Also read and apply settings when parsing and replaying dumps.

The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
  AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
  audioproc_f.

Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 11:40:28 +00:00
Sergey Silkin
042661b404 Revert "Frame rate controller per spatial layer."
This reverts commit ae9e188e67a489db597224e3cfcfdee04edf0cba.

Reason for revert: Verify if this causes chromium:882358.

Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}

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

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

Bug: webrtc:9682, chromium:882358
Change-Id: Idc4051eef72104823038ed9139bb9c75018f7d86
Reviewed-on: https://webrtc-review.googlesource.com/99082
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24646}
2018-09-10 11:24:33 +00:00
Erik Språng
fb2a66a58a libvpx vp8 encoder: get frame drop setting from temporal layer
Today, the internal frame dropper in libvpx vp8 encoder is enabled or
disabled based on video or screen content. This is then expected to
match up with screenshare vs default temporal layers implementation.

This cl makes libvpx query the temporal layers implementation as well,
breaking this implicit dependency and allows frames to be dropped if
default temporal layers is used with screen content.

Bug: webrtc:9734
Change-Id: If2523a211f4929f16e65a02fa7a6b4edf7328571
Reviewed-on: https://webrtc-review.googlesource.com/99062
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24645}
2018-09-10 11:10:11 +00:00
philipel
0417eadbf2 Removed unused member |last_unwrap_| from RtpFrameReferenceFinder.
Bug: none
Change-Id: Ideb876d89dbab7a9f4c8c46d95217f00e07b62d6
Reviewed-on: https://webrtc-review.googlesource.com/98862
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24644}
2018-09-10 10:03:41 +00:00
Ivo Creusen
4384f53285 Add more useful information to NetEqState and implement action_times_ms
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.

Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
2018-09-10 09:10:53 +00:00
Mirko Bonadei
76dac9ac2f Fix no_global_constructors in modules/video_capture.
Bug: webrtc:9693
Change-Id: Ia917ab824f18991cfdcffa04ea9c063c6a224532
Reviewed-on: https://webrtc-review.googlesource.com/98640
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24641}
2018-09-10 09:03:39 +00:00
Danil Chapovalov
0b445c6271 Cleanup RtpPacketizerVp9
Merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If230a4ea901b5cdd6a376f8dd2db48e94d6dca36
Reviewed-on: https://webrtc-review.googlesource.com/98866
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24635}
2018-09-07 22:01:46 +00:00
Danil Chapovalov
792df6b4b9 Make RtcpTransceiver destructor non-blocking
At cost of removing assumption callbacks can't be used after destructor.

Bug: webrtc:8239
Change-Id: Id79f7553528cf6c102d3ee0bf7aa2de5b0437d2a
Reviewed-on: https://webrtc-review.googlesource.com/98860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24632}
2018-09-07 15:34:08 +00:00
Mirko Bonadei
d7027dc081 Revert "Fix no_global_constructors in audio_processing/agc2/rnn_vad."
This reverts commit 5e2e66d8a0fd5e1bf9b3efc54a94cba3e7088b00.

Reason for revert: Change implementation.

Original change's description:
> Fix no_global_constructors in audio_processing/agc2/rnn_vad.
> 
> Bug: webrtc:9693
> Change-Id: Ica997d5cbe28288720325a51058a40a37c612665
> Reviewed-on: https://webrtc-review.googlesource.com/98583
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24617}

TBR=mbonadei@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org

Change-Id: I9e30f6ec08baa22a8d6c15546341000738c095b6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9693
Reviewed-on: https://webrtc-review.googlesource.com/98842
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24631}
2018-09-07 13:34:39 +00:00
Philip Eliasson
d52a1a6971 Reland "Remove RTPVideoHeader::vp8() accessors."
This reverts commit 1811c04f22a26da3ed2832373a5c92a9786420c3.

Reason for revert: Downstream projects fixed.

Original change's description:
> Revert "Remove RTPVideoHeader::vp8() accessors."
> 
> This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.
> 
> Reason for revert: Break downstream projects.
> 
> Original change's description:
> > Remove RTPVideoHeader::vp8() accessors.
> > 
> > Bug: none
> > Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> > Reviewed-on: https://webrtc-review.googlesource.com/93321
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24626}
> 
> TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com
> 
> Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: none
> Reviewed-on: https://webrtc-review.googlesource.com/98864
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24628}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I9246f36e638108ae4fc46c1ae4559c8205d50fc1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24629}
2018-09-07 13:04:07 +00:00
Philip Eliasson
1811c04f22 Revert "Remove RTPVideoHeader::vp8() accessors."
This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.

Reason for revert: Break downstream projects.

Original change's description:
> Remove RTPVideoHeader::vp8() accessors.
> 
> Bug: none
> Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> Reviewed-on: https://webrtc-review.googlesource.com/93321
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24626}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98864
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24628}
2018-09-07 12:36:17 +00:00
philipel
5470f4030f Renamed GetCodecHeader to GetRtpVideoHeader in RtpFrameObject.
Bug: none
Change-Id: I158a19dc85ef12dc86f603ff0f6618b89cb1c242
Reviewed-on: https://webrtc-review.googlesource.com/98863
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24627}
2018-09-07 12:35:12 +00:00
philipel
af7afc6642 Remove RTPVideoHeader::vp8() accessors.
Bug: none
Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
Reviewed-on: https://webrtc-review.googlesource.com/93321
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24626}
2018-09-07 12:01:19 +00:00
Sergey Silkin
2f864fb4ab Handle empty GOF.
Assume that stream has single temporal layer if number of frames in GOF
is set to zero (valid case).

Bug: chromium:879584
Change-Id: I7ced082190e40c1bf4cc1468babfd98b0a61f0dd
Reviewed-on: https://webrtc-review.googlesource.com/98800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24622}
2018-09-07 10:19:06 +00:00
Niels Möller
fe3240aeae Reland "Delete class EventTimerWrapper."
This is a reland of a421775a6d4f78f7aa9c3ea020a8834e049efbcc

Original change's description:
> Delete class EventTimerWrapper.
>
> Only user, iSACTest, refactored to use a sleep instead.
>
> Bug: webrtc:3380
> Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
> Reviewed-on: https://webrtc-review.googlesource.com/96802
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24541}

Tbr: henrik.lundin@webrtc.org
Bug: webrtc:3380
Change-Id: I541473b9c3ce2020f76d420598a7b10766f1d2a9
Reviewed-on: https://webrtc-review.googlesource.com/98481
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24620}
2018-09-07 09:54:55 +00:00
Danil Chapovalov
fa5ec8d20d Use signed integers for limiting packet size in video packetizers
Using signed integers allows to centralize checking of edge cases
in RtpPacketizer::SplitAboutEqually and
reduce chance of hitting issues with size_t underflow

Bug: webrtc:9680
Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08
Reviewed-on: https://webrtc-review.googlesource.com/98602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24618}
2018-09-07 09:24:18 +00:00
Mirko Bonadei
5e2e66d8a0 Fix no_global_constructors in audio_processing/agc2/rnn_vad.
Bug: webrtc:9693
Change-Id: Ica997d5cbe28288720325a51058a40a37c612665
Reviewed-on: https://webrtc-review.googlesource.com/98583
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24617}
2018-09-07 08:08:45 +00:00
Emircan Uysaler
ed425d12c4 Multiplex codec cleanups
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.

Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
2018-09-06 23:29:15 +00:00
Per Åhgren
6a4fd19bbd AEC3: Parametrize the delay estimator to leverage strong echo paths
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.

To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.

Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
2018-09-06 23:01:58 +00:00
Mirko Bonadei
c003a1f067 Fix no_global_constructors in congestion_controller/pcc.
Bug: webrtc:9693
Change-Id: I7d654656b4f350c120d25dcc0f66541f47ccc919
Reviewed-on: https://webrtc-review.googlesource.com/98582
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24609}
2018-09-06 16:18:19 +00:00