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Author SHA1 Message Date
Sebastian Jansson
011d3a125e Revert "Cleanup in NetworkEmulationManagerImpl." 
This reverts commit aa274d028710cb9d8f1fe3fe5ff68fe7b04e4cf2.

Reason for revert: Depends on reverted CL.

Original change's description:
> Cleanup in NetworkEmulationManagerImpl.
> 
> Bug: webrtc:9883
> Change-Id: I845ecd66ceda9eb8c144e4c10ff374c050413a04
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133344
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27664}

TBR=srte@webrtc.org,titovartem@webrtc.org

Change-Id: Ib62f19e382ee0a43f5892fcc694395adb410c36f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133484
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27673}
 2019-04-17 15:57:01 +00:00
Danil Chapovalov
28de7cfd64 Use DefaultTaskQueueFactory instead of Global in test RealTimeController 
Bug: webrtc:10284
Change-Id: I1c285078a9aae42ca811144f42066980a7b155fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132782
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27672}
 2019-04-17 15:52:28 +00:00
chromium-webrtc-autoroll
899280c485 Roll chromium_revision 3904a4338b..14c052d7be (651612:651715) 
Change log: 3904a4338b..14c052d7be
Full diff: 3904a4338b..14c052d7be

Changed dependencies
* src/base: 8b6a1456f6..9f11811ffe
* src/build: a0b2e3b270..1840a3c7e4
* src/ios: b25ad32c5a..bd0a450991
* src/testing: e005720bc1..fac08fe808
* src/third_party: a5fa03ace1..80de4cb2ff
* src/third_party/depot_tools: 7e7523be4e..e72279d25f
* src/tools: ba9616c509..bc2401b292
DEPS diff: 3904a4338b..14c052d7be/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I28b58be0971afce55119b5e657a4c3e4349782e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133520
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27671}
 2019-04-17 15:48:03 +00:00
Daniel Lee
63658d06ec Revert "Ensure that we always set values for min and max audio bitrate." 
This reverts commit e47aee3b864fe5a4f964d405a7f6f3ac8c49f174.

Reason for revert: Breaks downstream project

Original change's description:
> Ensure that we always set values for min and max audio bitrate.
> 
> Use (in order from lowest to highest precedence):
> -- fixed 32000bps
> -- fixed target bitrate from codec
> -- explicit values from the rtp encoding parameters
> -- Final precedence is given to field trial values from
>    WebRTC-Audio-Allocation
> 
> Bug: webrtc:10487
> Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
> Reviewed-by: Minyue Li <minyue@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Daniel Lee <dklee@google.com>
> Cr-Commit-Position: refs/heads/master@{#27667}

TBR=solenberg@webrtc.org,stefan@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,minyue@webrtc.org,minyue@google.com,dklee@google.com

Change-Id: Ie975cf40e65105d1e4cfab417b220b6bfc34592b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27670}
 2019-04-17 15:47:00 +00:00
Artem Titarenko
b4b53bd8cc Revert "Only process cross traffic simulation if added." 
This reverts commit 491d29448e4a0f501f1ea115b218ed5250fbd3dc.

Reason for revert: Breaks internal iOS waterfall

Original change's description:
> Only process cross traffic simulation if added.
> 
> This avoids extra processing overhead when there's no cross traffic
> simulation active. To allow the use of SendTask, GlobalTimeController
> is adjusted so it always overrides yield behavior.
> 
> Also adding sequence checkers to protect against races on
> read access.
> 
> Bug: webrtc:10365
> Change-Id: I55c6ceb22f36ec19a4fca48cff500905192d1a16
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133167
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27658}

TBR=nisse@webrtc.org,srte@webrtc.org,titovartem@webrtc.org

Change-Id: I3b176b0ec202a047c4436e3c2136e8170faea41e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133482
Reviewed-by: Artem Titarenko <artit@google.com>
Commit-Queue: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27669}
 2019-04-17 15:39:10 +00:00
Sebastian Jansson
40889f35fc Removes TimeMicros interface from ThreadProcessingFakeClock. 
Bug: webrtc:9883
Change-Id: Ib48872f81f734b09e3ffa4d9d26da79177b02303
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133341
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27668}
 2019-04-17 15:37:48 +00:00
Daniel Lee
e47aee3b86 Ensure that we always set values for min and max audio bitrate. 
Use (in order from lowest to highest precedence):
-- fixed 32000bps
-- fixed target bitrate from codec
-- explicit values from the rtp encoding parameters
-- Final precedence is given to field trial values from
   WebRTC-Audio-Allocation

Bug: webrtc:10487
Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27667}
 2019-04-17 14:40:23 +00:00
Sebastian Jansson
d624c3903c Make MessageQueue processing an optional feature of FakeClock 
This is used to avoid thread processing in simulated time
controller. This saves up to 30% execution time in debug builds.

Bug: webrtc:10365
Change-Id: Ie83dfb2468d371e4687d28c776acf7e23eb411d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133173
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27666}
 2019-04-17 14:10:16 +00:00
Mirta Dvornicic
8a7dcb163e Prepare to move SPS VUI rewriting out of H.264 packetizer. 
- add ParseOutgoingBitstreamAndRewriteSps to SpsVuiRewriter
  which takes encoded H.264 bitstream and NAL unit boundaries,
  rewrites SPS if needed and updates the NAL unit boundaries
  accordingly
- move SPS rewriting stats updates to SpsVuiRewriter

Bug: webrtc:10559
Change-Id: I7ca21756628ee6d6abbcbd501bdb4f3df024168b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133174
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27665}
 2019-04-17 13:21:30 +00:00
Sebastian Jansson
aa274d0287 Cleanup in NetworkEmulationManagerImpl. 
Bug: webrtc:9883
Change-Id: I845ecd66ceda9eb8c144e4c10ff374c050413a04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133344
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27664}
 2019-04-17 13:18:10 +00:00
Henrik Boström
cf96e0f87d Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. 
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent

These are already existed in StreamDataCounters. This CL takes care of
the plumbing of these values to the standard stats collector.

TBR=solenberg@webrtc.org

Bug: webrtc:10447
Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27663}
 2019-04-17 13:04:50 +00:00
Niels Möller
decc07679d Expose send parameters for RemoteEstimatorProxy for field trial. 
Bug: None
Change-Id: I14c74f55ed534fff19042423dad63641bd0fa047
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133187
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27662}
 2019-04-17 12:45:47 +00:00
Per Kjellander
416d5db75d Add field trial to AimdRateController to only increase while not in ALR 
The idea is that when ALR is detected, the encoder can not produce the bitrate
needed for the delay based estimator to detect overuse and thus the delay based
estimator should not be allowed to increase further.
Likewise, if ALR is not detected, the delay based estimator is allowed to
increase the BWE to ensure that there is no region where the BWE can get stuck.

BUG=webrtc:10542

Change-Id: Ic94b708461c9077fd09132ee4ecb6279ffcd5f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133190
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27661}
 2019-04-17 12:05:24 +00:00
Elad Alon
979c4426a4 Rename "UpdateLayerConfig" to "NextFrameConfig" 
Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.

Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
 2019-04-17 11:59:07 +00:00
Elad Alon
123ee9be8f OnLossNotification() receives references 
A typo in a previous CL made OnLossNotification() accept its
single argument as a const-value, rather than a const-reference.

Bug: webrtc:10501
Change-Id: I5e6f9c79f15205b75ec90a53d3fccf3dd9927e33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27659}
 2019-04-17 11:47:25 +00:00
Sebastian Jansson
491d29448e Only process cross traffic simulation if added. 
This avoids extra processing overhead when there's no cross traffic
simulation active. To allow the use of SendTask, GlobalTimeController
is adjusted so it always overrides yield behavior.

Also adding sequence checkers to protect against races on
read access.

Bug: webrtc:10365
Change-Id: I55c6ceb22f36ec19a4fca48cff500905192d1a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133167
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27658}
 2019-04-17 10:59:06 +00:00
Niels Möller
c29fa1bf67 Drop usage of legacy VCMPacketRequestCallback 
It's called from VideoReceiver::Process, which we no longer use.

Bug: webrtc:7408
Change-Id: I2ce5b7a07437e110d20d04aa159dddf245504abe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133189
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27657}
 2019-04-17 10:14:06 +00:00
Niels Möller
bbdf0f67b3 Don't call VideoReceiver::SetNackSettings 
It configures features of the old jitter buffer which we don't use.

Bug: webrtc:7408
Change-Id: I0f0e9c20b76fb5589f12bc696022d7d2ffef0ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133168
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27656}
 2019-04-17 08:52:20 +00:00
chromium-webrtc-autoroll
6f5e84894e Roll chromium_revision cd3b1fea2d..3904a4338b (651505:651612) 
Change log: cd3b1fea2d..3904a4338b
Full diff: cd3b1fea2d..3904a4338b

Changed dependencies
* src/build: 57582c144e..a0b2e3b270
* src/ios: d7fe686bf5..b25ad32c5a
* src/testing: 8a394c7d20..e005720bc1
* src/third_party: 2dd0dc0f3a..a5fa03ace1
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/73185de866..acbf095c15
* src/tools: 0e2ddb5b1a..ba9616c509
DEPS diff: cd3b1fea2d..3904a4338b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I2db107498264a63d5a2746c934a84bd8318add0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133423
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27655}
 2019-04-17 06:36:14 +00:00
Jonas Oreland
a3aa9bd75b Make VideoBitrateAllocatorFactory injectable. 
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).

With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.

WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@

Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
 2019-04-17 06:17:34 +00:00
chromium-webrtc-autoroll
a1a93fba36 Roll chromium_revision 1645d3d94d..cd3b1fea2d (651391:651505) 
Change log: 1645d3d94d..cd3b1fea2d
Full diff: 1645d3d94d..cd3b1fea2d

Changed dependencies
* src/base: fc5b2dc0e5..8b6a1456f6
* src/build: 1daeec1375..57582c144e
* src/ios: 29d29b7f20..d7fe686bf5
* src/third_party: 6006d39fc9..2dd0dc0f3a
* src/third_party/depot_tools: 4b79c3870c..7e7523be4e
* src/tools: fc26e4ab83..0e2ddb5b1a
DEPS diff: 1645d3d94d..cd3b1fea2d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: If457a98d7e246e66989d2c6ebda7a89dff8c3835
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133400
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27653}
 2019-04-16 23:34:50 +00:00
Seth Hampson
1859dc04fd Revert "Refactoring DataContentDescription class" 
This reverts commit 8a9193c217d818fea77b9540bd4ca7ebad53db76.

Reason for revert: Breaks downstreams

Original change's description:
> Refactoring DataContentDescription class
> 
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::DataContentDescription (used for RTP data) and
> cricket::SctpDataContentDescription (used for SCTP only).
> 
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
> 
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> 
> Bug: webrtc:10358
> Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27651}

TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I3b8a68cd481c41ce30eeb5ffbc5da735a9659019
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133360
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27652}
 2019-04-16 20:56:06 +00:00
Harald Alvestrand
8a9193c217 Refactoring DataContentDescription class 
This CL splits the cricket::DataContentDescription class into
two classes: cricket::DataContentDescription (used for RTP data) and
cricket::SctpDataContentDescription (used for SCTP only).

SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".

Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#

Bug: webrtc:10358
Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27651}
 2019-04-16 20:34:34 +00:00
chromium-webrtc-autoroll
5111c338ca Roll chromium_revision 2e4f1b8087..1645d3d94d (650971:651391) 
Change log: 2e4f1b8087..1645d3d94d
Full diff: 2e4f1b8087..1645d3d94d

Changed dependencies
* src/base: e92dbd2eed..fc5b2dc0e5
* src/build: 45887bbd00..1daeec1375
* src/ios: e9e306553a..29d29b7f20
* src/testing: c044935b34..8a394c7d20
* src/third_party: 082b5dec62..6006d39fc9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9de7d78395..73185de866
* src/third_party/depot_tools: db58954c8c..4b79c3870c
* src/third_party/libvpx/source/libvpx: c46694c1d9..da5be113f3
* src/tools: 97410bd377..fc26e4ab83
DEPS diff: 2e4f1b8087..1645d3d94d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org, jianj@chromium.org,
BUG=None

Change-Id: Ibc5bb17e0663ea54c33630086d4ea8b1624ae296
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133321
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27650}
 2019-04-16 19:39:44 +00:00
Artem Titov
153056b059 Add ability to play audio in circle for TestAudioDevice wav file capturer 
Also use this ability in PC smoke test.

Bug: webrtc:10138
Change-Id: I83d526344f203082a19377d9642c9e453454f7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133163
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27649}
 2019-04-16 15:33:03 +00:00
Niels Möller
50150a18f4 Delete VCMJitterBuffer::UpdateRtt and related unused code 
Bug: webrtc:7408
Change-Id: Ic4bd745ed36dd7f7d5eec6531e05204035a545fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133188
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27648}
 2019-04-16 15:13:13 +00:00
Rasmus Brandt
034f1916b5 Revert "Add more adaptation text logs for VideoSendStream." 
This reverts commit d0298f4b161fe2afd4c6b334da31218c115a7eeb.

Reason for revert: This change makes the text output for three simulcast layers dangerously close to 1024 characters. Will reland with a larger value.

Original change's description:
> Add more adaptation text logs for VideoSendStream.
> 
> Tested: Manual tests in app.
> Bug: None
> Change-Id: I2739a23d37c05cbe1ba9be5c788d1c647265a895
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133186
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27644}

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

Change-Id: Ic6c0c863f6e4aba12feb6c6938db2930396c32f4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133204
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27647}
 2019-04-16 14:46:57 +00:00
Artem Titov
dd1c16f00c Use absl::make_unique in TestAudioDeviceModule factory methods 
Bug: webrtc:10138
Change-Id: Ibe9f4b4343b8e5c9a5e1a6d41bd06b24d69db878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133166
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27646}
 2019-04-16 14:43:55 +00:00
Niels Möller
7eb4248b02 Delete unused class Vp9SsMap 
Appears unused since https://codereview.webrtc.org/1426813002

Bug: webrtc:7408
Change-Id: I13b8c35548314705eaf31ff2366636f358fef718
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133203
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27645}
 2019-04-16 14:36:55 +00:00
Rasmus Brandt
d0298f4b16 Add more adaptation text logs for VideoSendStream. 
Tested: Manual tests in app.
Bug: None
Change-Id: I2739a23d37c05cbe1ba9be5c788d1c647265a895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133186
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27644}
 2019-04-16 13:52:53 +00:00
Minyue Li
7d204d5ce9 Disallow buffer level filtering for DTX packets. 
We knew that we should not update buffer level during DTX period. We already fulfill this upon no packet receipt. But we missed doing it for DTX-signaling packets. This CL is to fix that.

Bug: b/129521878
Change-Id: I72ca18e3b21e956123fe6e3119ef0d7c981c9eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133183
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27643}
 2019-04-16 13:31:32 +00:00
Per Kjellander
494947bbcf Remove direct use of FieldTrials from modules/remote_bitrate_estimator 
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig

BUG=webrtc:10335

Change-Id: Ie148cb466f86d8fa1ded5c7f125fbcccf6e7dbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132714
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27642}
 2019-04-16 13:23:12 +00:00
Konrad Hofbauer
bfb735b5a1 Exclude RTX from pacer delay plots 
This is because padding (sent as RTX) makes the plot unreadable.

TBR=terelius@webrtc.org

Bug: None
Change-Id: Iddf681eab6ec826c6f3c620aac65e2bd6f31b895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133182
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27641}
 2019-04-16 13:05:52 +00:00
Guido Urdaneta
70c2db1aa0 Reland "Make negotiationneeded processing in PeerConnection spec compliant." 
The new processing applies only in Unified Plan mode.
Plan B retains the old-style processing.

This is a reland of 1fa06041bcd8a0119e557d16e7b54a9110c5ad03

Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}

Bug: chromium:740501
Change-Id: I048ae81b2b00086f6d669e94eecf426f0db0ec08
TBR: steveanton@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133162
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27640}
 2019-04-16 12:04:33 +00:00
Elad Alon
c3f31dc395 Avoid copying of InlinedVector by DefaultTemporalLayers ctor 
Bug: None
Change-Id: I31767a97ad5438001b6fc348157ef25a97ed6168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27639}
 2019-04-16 10:50:41 +00:00
Sebastian Jansson
a7e78f24dd Default enabling decoding on task queue. 
Bug: webrtc:10365
Change-Id: I3e367129317e7a7f7f1e1531d7013e07ca5101bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133160
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27638}
 2019-04-16 09:53:34 +00:00
Sebastian Jansson
eb02a76b36 Allow negative delta in EventRateCounter. 
Bug: webrtc:10365
Change-Id: I13192d1e179ef5637210cfea3e67010d00117957
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133025
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27637}
 2019-04-16 09:32:42 +00:00
Niels Möller
6cf61f53ad Delete unneeded includes of async_invoker.h 
Bug: None
Change-Id: I3753592f8eb53eb2b31cf645b80c446bd2251404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133027
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27636}
 2019-04-16 07:28:06 +00:00
Sebastian Jansson
a4c22b9662 Using NetworkEmulationManager in Scenario tests. 
Bug: webrtc:9510
Change-Id: Ib619526269c58f0c46c0c1f01ba6c0efa5f79ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27635}
 2019-04-16 06:24:26 +00:00
Steve Anton
884adca3a0 Log details when RtpDemuxer fails to deliver a packet 
Bug: None
Change-Id: Ie9dc5c3c545073d2e43b464d2585cb945eb346fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131360
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27634}
 2019-04-16 00:47:53 +00:00
Amit Hilbuch
6df49d8983 Fixing issue with creating StreamParams when track id is not signaled. 
Current logic requires a stream id and track ids when creating a stream
that does not have SSRCs signaled.
This change removes the requirement for stream ids. The requirement for
track id is softer, as one should be generated when it is not present.

Bug: webrtc:10551
Change-Id: Ibc0cc181c6b18efa8394b6c0e4820e3a13da70c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27633}
 2019-04-16 00:34:04 +00:00
chromium-webrtc-autoroll
53625ca8ab Roll chromium_revision a35784fb06..2e4f1b8087 (650856:650971) 
Change log: a35784fb06..2e4f1b8087
Full diff: a35784fb06..2e4f1b8087

Changed dependencies
* src/base: 8beae815b0..e92dbd2eed
* src/build: 7c4aed4851..45887bbd00
* src/buildtools: 218cb3d12e..d5c58b84d5
* src/buildtools/third_party/libc++/trunk: fbddc46986..9b96c3dbd4
* src/ios: 468d16d887..e9e306553a
* src/third_party: 1b3f717046..082b5dec62
* src/third_party/r8: SlcbUnEufAQ-iuOwGOl8yYQuctmpf7bMqh59kBfpil0C..BReCwfbVwCNM2Ry4QpnrwlE3Y5gPJ2rRoyMbxFS0-4UC
* src/tools: ff070cdf4a..97410bd377
DEPS diff: a35784fb06..2e4f1b8087/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ie44fc059c61e2dbb26be7508ab9de53c940e25f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133017
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27632}
 2019-04-15 21:50:51 +00:00
Sebastian Jansson
5d97f552ba Allow injection of time controller to NetworkEmulationManagerImpl. 
Bug: webrtc:10365
Change-Id: I6a0e04459f75e8134787e605057dcb03cae55cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132780
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27631}
 2019-04-15 18:53:24 +00:00
chromium-webrtc-autoroll
2f92b414ae Roll chromium_revision 8d55ca9363..a35784fb06 (650742:650856) 
Change log: 8d55ca9363..a35784fb06
Full diff: 8d55ca9363..a35784fb06

Changed dependencies
* src/base: 39249a7fe2..8beae815b0
* src/build: 3b075157b4..7c4aed4851
* src/ios: 1fa5b61040..468d16d887
* src/third_party: dd3857ca4d..1b3f717046
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b614c567e8..9de7d78395
* src/tools: dc8f3b2879..ff070cdf4a
DEPS diff: 8d55ca9363..a35784fb06/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Icfcd4f6a1b83e903d0907294c1f282d6a80b17fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133061
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#27630}
 2019-04-15 18:31:24 +00:00
Gustaf Ullberg
2bab5ad3b1 AEC3: Avoid using filter output in suppression gain computation in non-linear mode 
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.

This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.

Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
 2019-04-15 16:08:41 +00:00
Henrik Boström
01738c63aa Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 
This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}
 2019-04-15 16:06:01 +00:00
Erik Språng
3d11e2f81c Allow encoder target bitrate to reach media rate if there is headroom. 
This CL adds a field trial that enables the EncoderBitrateAdjuster to
allow higher target bitrate if we are not network constrained. We still
don't allow the bitrate to go higher than the average target media rate
though.

Bug: webrtc:10155
Change-Id: Id5995070aa0cbe84b9305a422279141b38664bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132717
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27627}
 2019-04-15 15:11:39 +00:00
Minyue Li
f9846bc172 Adding DTX logic to FakeDecodeFromFile (used be NetEqTest). 
Bug: b/129521878
Change-Id: Ifcf868048a39ef1d2cc736988479f921e668167b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132799
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27626}
 2019-04-15 15:03:39 +00:00
Sebastian Jansson
72b7524d87 Adds more stats to CallStatsCollector. 
Also adding checks to avoid adding empty stats.

Bug: webrtc:10365
Change-Id: I37ab32a3d4271fcad419f17841a8d2e524d73245
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133020
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27625}
 2019-04-15 14:47:56 +00:00
Sebastian Jansson
df88cc014a Allow injection of network estimator into GoogCC. 
Bug: webrtc:10498
Change-Id: Ie9225411db201dfcfa0a37a3c40992acbdc215bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27624}
 2019-04-15 14:12:08 +00:00