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Author SHA1 Message Date
Steve Anton
ae226f65c8 Use Abseil container algorithms in p2p/ 
Bug: None
Change-Id: I02dd19efa201bd9d55d0f7c2e1496693017a6848
Reviewed-on: https://webrtc-review.googlesource.com/c/120001
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26455}
 2019-01-29 21:52:18 +00:00
Piotr (Peter) Slatala
3e659b811a Remove deprecated OnKeyFrame method from video sink interface in media transport 
Bug: webrtc:9719
Change-Id: I0d172e41bfe46ae4eec25de0e20f2ca4bfc64c19
Reviewed-on: https://webrtc-review.googlesource.com/c/120420
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26454}
 2019-01-29 21:27:52 +00:00
chromium-webrtc-autoroll
9057260777 Roll chromium_revision bf03673fd1..241ac98bfc (626985:627089) 
Change log: bf03673fd1..241ac98bfc
Full diff: bf03673fd1..241ac98bfc

Changed dependencies
* src/base: fc19b2cdf9..b75c898994
* src/ios: 8214c6c2d8..035b561e5c
* src/testing: 80efe67905..ee6e9ab571
* src/third_party: 818b97d2b4..f20a194062
* src/third_party/harfbuzz-ng/src: 36fb2b4da9..fe53292310
* src/tools: f2908e7a0d..723be0ba24
DEPS diff: bf03673fd1..241ac98bfc/DEPS

No update to Clang.

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

Change-Id: I5b240ab701d0ce5bd6ac35a9abf70dafc0c7a380
Reviewed-on: https://webrtc-review.googlesource.com/c/120442
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@{#26453}
 2019-01-29 19:32:44 +00:00
Erik Språng
5118bbc8b7 Add ability to set max probing bitrate via GoogCcNetworkController 
Bug: webrtc:10223
Change-Id: I8e9ee0cd333634e7d0b53d3d446a580374cc88b4
Reviewed-on: https://webrtc-review.googlesource.com/c/120342
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26452}
 2019-01-29 19:19:04 +00:00
Zach Stein
d3be0171b0 Remove unused PacketLossEstimator class 
These metrics were never hooked up to anything.

Bug: webrtc:7028
Change-Id: Id6fdf146de615839820f7ad3805eb42450c76c21
Reviewed-on: https://webrtc-review.googlesource.com/c/120303
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26451}
 2019-01-29 19:03:24 +00:00
Sebastian Jansson
8c8feb9d2b Moves packet overhead from network nodes to simulation. 
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)

Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
 2019-01-29 16:55:04 +00:00
Florent Castelli
c1a0bcbe89 Implement the encoding RtpParameter scaleResolutionDownBy 
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.

Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
 2019-01-29 14:32:17 +00:00
Elad Alon
411b49be17 Break FrameConfig out of Vp8TemporalLayers 
FrameConfig is not specific to temporal layers. Anything that
can control referenced/updated buffers could potentially use it.

Bug: webrtc:10259
Change-Id: I04ed177ee884693798c3b69e35fd4255ce1e9062
Reviewed-on: https://webrtc-review.googlesource.com/c/120355
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26448}
 2019-01-29 14:13:55 +00:00
chromium-webrtc-autoroll
31a739e90b Roll chromium_revision 531da0eda2..bf03673fd1 (626885:626985) 
Change log: 531da0eda2..bf03673fd1
Full diff: 531da0eda2..bf03673fd1

Changed dependencies
* src/base: efcb688da3..fc19b2cdf9
* src/build: 4ab9949ff1..9dbdd5c2ae
* src/testing: f4d07548ac..80efe67905
* src/third_party: 8f4dd7aebe..818b97d2b4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/eae881c2a8..a08f0fce79
* src/third_party/depot_tools: 3f812d07b2..b69515579d
* src/tools: baf934767b..f2908e7a0d
DEPS diff: 531da0eda2..bf03673fd1/DEPS

No update to Clang.

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

Change-Id: Ibb4584e9957e232ae8e68acb92ba91f4f2aca530
Reviewed-on: https://webrtc-review.googlesource.com/c/120363
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@{#26447}
 2019-01-29 13:37:54 +00:00
Alex Loiko
b4977de306 Receive-side ready for multiple channels. 
Made path from NetEq to AudioTransport ready for many-channel audio.
If there is one stream, we can handle anything that fits in an
AudioFrame. For many streams, the current limit is 6.

Some multi-channel combinations are not supported: e.g. if we get
stereo audio and attempt to play out 6 channels.

Changes:
* AudioFrameOperations - replaced the MonoTo* and *ToMono methods by
  UpmixChannels & DownmixChannels.
* AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing
  algorithm to handle many channels.

Bug: webrtc:8649
Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106040
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26446}
 2019-01-29 12:43:23 +00:00
Alex Loiko
7a3e43a5d7 Reland of Opus multistream. 
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/111750.

This time we don't use the multistream decoder unless we have to.
(Which is when #channels >2). Pros: don't make downstream projects
crash due to used up stack space, a few % more efficiency for the
typical case (because multistream adds some overhead). Cons: Messy
C-code with "union" types and #define MACROs, probably more
maintenance.

Bug: webrtc:8649
Change-Id: I4253a5e0c382f67ac7c6731dc6602a31e6779e63
Reviewed-on: https://webrtc-review.googlesource.com/c/120049
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26445}
 2019-01-29 12:16:19 +00:00
Jesús de Vicente Peña
e5ccf5fe5b APM: adding a missing header when dumping files in APM 
Change-Id: Ife8d45179354a1dd7525175e11a6016af2777910
Bug: webrtc:10255
Reviewed-on: https://webrtc-review.googlesource.com/c/120345
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26444}
 2019-01-29 11:32:20 +00:00
Gustaf Ullberg
68d6d44197 AEC3: Remove remaining kill-switches 
This CL concludes the post-launch removal of kill-switches is AEC3.

Kill-switches removed:
WebRTC-Aec3AdaptErleOnLowRenderKillSwitch
WebRTC-Aec3AgcGainChangeResponseKillSwitch
WebRTC-Aec3BoundedNearendKillSwitch
WebRTC-Aec3EarlyShadowFilterJumpstartKillSwitch
WebRTC-Aec3EnableAdaptiveEchoReverbEstimation
WebRTC-Aec3EnforceSkewHysteresis1
WebRTC-Aec3EnforceSkewHysteresis2
WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch
WebRTC-Aec3MisadjustmentEstimatorKillSwitch
WebRTC-Aec3OverrideEchoPathGainKillSwitch
WebRTC-Aec3RapidAgcGainRecoveryKillSwitch
WebRTC-Aec3ResetErleAtGainChangesKillSwitch
WebRTC-Aec3ShadowFilterBoostedJumpstartKillSwitch
WebRTC-Aec3ShadowFilterJumpstartKillSwitch
WebRTC-Aec3SmoothSignalTransitionsKillSwitch
WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch
WebRTC-Aec3SoftTransparentModeKillSwitch
WebRTC-Aec3StandardNonlinearReverbModelKillSwitch
WebRTC-Aec3StrictDivergenceCheckKillSwitch
WebRTC-Aec3UseOffsetBlocks
WebRTC-Aec3UseStationarityPropertiesKillSwitch
WebRTC-Aec3UtilizeShadowFilterOutputKillSwitch
WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch
WebRTC-Aec3FilterQualityStateKillSwitch
WebRTC-Aec3NewSaturationBehaviorKillSwitch
WebRTC-Aec3GainLimiterDeactivationKillSwitch
WebRTC-Aec3EnableErleUpdatesDuringReverbKillSwitch

The change has been tested for bit-exactness.

Bug: webrtc:8671
Change-Id: I42816b9d1c875cec0347034c6e2ed4ff5db6ec0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119942
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26443}
 2019-01-29 10:31:45 +00:00
Mirko Bonadei
649a4c2ea3 [clang-tidy] Apply performance-inefficient-vector-operation fixes. 
This CL applies clang-tidy's performance-inefficient-vector-operation
[1] on the WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-inefficient-vector-operation.html

Bug: webrtc:10252
Change-Id: I824caab2a5746036852e00d714b89aa5ec030ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/120052
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26442}
 2019-01-29 09:45:21 +00:00
Niels Möller
949f0fdc10 Move FrameCountObserver from RTPSender to RtpVideoSender 
Tbr: sprang@webrtc.org # Trivial change to rtp_video_stream_receiver.cc
Bug: webrtc:7135
Change-Id: Ic292fb02046ea800d7f0876900997d96ed0099d6
Reviewed-on: https://webrtc-review.googlesource.com/c/120161
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26441}
 2019-01-29 09:31:11 +00:00
Oleh Prypin
3e8b7e9b6b mb: remove 'type': 'gn' because it's the default and doesn't mean anything 
Bug: None
Change-Id: Ib987f180e48d42678d4924079281010279292297
Reviewed-on: https://webrtc-review.googlesource.com/c/120341
Commit-Queue: Oleh Prypin <oprypin@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26440}
 2019-01-29 09:29:41 +00:00
Elad Alon
e008248c7d Only instantiate TemporalLayersChecker in debug builds 
Bug: None
Change-Id: I0f700451df4c9adfc07c77e62a5964c85079fefa
Reviewed-on: https://webrtc-review.googlesource.com/c/120051
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26439}
 2019-01-29 09:01:18 +00:00
Elad Alon
f5b216a1b7 Pass explicit frame dependency information to RtpPayloadParams 
Prior to this CL, RtpPayloadParams had code that assumed
dependency patterns in VP8, in order to write that information
into the [Generic Frame Descriptor] RTP extension.

This CL starts moving that code out of RtpPayloadParams.
Upcoming CLs will migrate additional encoder-wrappers to
the new scheme, then remove the deprecated code.

Bug: webrtc:10249
Change-Id: I5fc84aedf8e11f79d52b989ff8b7ce9568b6cf32
Reviewed-on: https://webrtc-review.googlesource.com/c/119958
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26438}
 2019-01-29 08:59:48 +00:00
Johnny Lee
7248b40344 Added VcmCapturer::Create loop to allow nonzero device index. 
Bug: webrtc:10181
Change-Id: I29c701ed756416b63d377e9b9137fffeba1f7f2e
Reviewed-on: https://webrtc-review.googlesource.com/c/116440
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26437}
 2019-01-29 08:06:22 +00:00
chromium-webrtc-autoroll
f7f227c6f9 Roll chromium_revision ed7fd9b77f..531da0eda2 (626752:626885) 
Change log: ed7fd9b77f..531da0eda2
Full diff: ed7fd9b77f..531da0eda2

Changed dependencies
* src/base: d185c046dc..efcb688da3
* src/build: 8d3f321ddb..4ab9949ff1
* src/ios: 031317d0c2..8214c6c2d8
* src/testing: aac1f41bd4..f4d07548ac
* src/third_party: 4f78be851d..8f4dd7aebe
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/556d7714fd..eae881c2a8
* src/third_party/depot_tools: b19e8dff15..3f812d07b2
* src/third_party/libFuzzer/src: ee7a5b85c7..6134addcf3
* src/tools: 91e4520c63..baf934767b
DEPS diff: ed7fd9b77f..531da0eda2/DEPS

No update to Clang.

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

Change-Id: I00dcaf7da881c8fde68ba810b8a71730a3978f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/120302
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@{#26436}
 2019-01-29 04:21:35 +00:00
Steve Anton
3d02384487 Fix inverted DCHECK conditional 
This fixes a regression added in
https://webrtc-review.googlesource.com/c/src/+/119862

Bug: None
Change-Id: Ica4157d63da502298a04a35f9ddb7e8b124902e0
Tbr: amithi@webrtc.org
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/120301
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26435}
 2019-01-29 04:14:35 +00:00
Steve Anton
2c9ebefb44 Use Abseil container algorithms in media/ 
Bug: None
Change-Id: I292e3401bbf19a66271dd5ef2b3ca4f8dcfd155d
Reviewed-on: https://webrtc-review.googlesource.com/c/120003
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26434}
 2019-01-29 02:35:50 +00:00
Steve Anton
64b626b03f Use Abseil container algorithms in pc/ 
Bug: None
Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e
Reviewed-on: https://webrtc-review.googlesource.com/c/119862
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26433}
 2019-01-29 02:33:50 +00:00
Amit Hilbuch
b7446ed257 Removing receive RIDs and Simulcast Layers. 
In the January 22nd 2019 WebRTC meeting it was agreed that an offer
for sending (or receiving) simulcast should only contain the RIDs
of the layers that are sent by the client.
This change removes the complexity that was added to support sending
and receiving the single layer (and RID) that are sent from the server.

Bug: webrtc:10076
Change-Id: I8bae1336d5cb8ba2f91c5b62332dc69e67ddfd47
Reviewed-on: https://webrtc-review.googlesource.com/c/120242
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26432}
 2019-01-29 00:54:26 +00:00
chromium-webrtc-autoroll
9bcf80a6e5 Roll chromium_revision fa9574f1d1..ed7fd9b77f (626644:626752) 
Change log: fa9574f1d1..ed7fd9b77f
Full diff: fa9574f1d1..ed7fd9b77f

Changed dependencies
* src/base: aaf74170f9..d185c046dc
* src/build: 5aa5d9d0dc..8d3f321ddb
* src/ios: 37a9132775..031317d0c2
* src/third_party: 9f2ff3c970..4f78be851d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/514fe3e70d..556d7714fd
* src/third_party/depot_tools: bdb1123726..b19e8dff15
* src/tools: 3cb5afca12..91e4520c63
DEPS diff: fa9574f1d1..ed7fd9b77f/DEPS

No update to Clang.

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

Change-Id: I2d32f0aa3ab02ccd3f99f1df4fb7bfd9083e492a
Reviewed-on: https://webrtc-review.googlesource.com/c/120260
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@{#26431}
 2019-01-28 23:16:50 +00:00
Bjorn Terelius
733e087e63 Ignore duplicated incoming RTCP packets in RTC event log parser. 
Bug: webrtc:8111
Change-Id: I1082ff66cac9c3744811713d686b3d7f85bd7584
Reviewed-on: https://webrtc-review.googlesource.com/c/120200
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26430}
 2019-01-28 20:38:38 +00:00
chromium-webrtc-autoroll
a75f618c83 Roll chromium_revision 0a788fbaed..fa9574f1d1 (626455:626644) 
Change log: 0a788fbaed..fa9574f1d1
Full diff: 0a788fbaed..fa9574f1d1

Changed dependencies
* src/base: 8889f1fcd9..aaf74170f9
* src/build: a041d21740..5aa5d9d0dc
* src/ios: f43b824a07..37a9132775
* src/testing: 6ed975ab13..aac1f41bd4
* src/third_party: 73ebf220db..9f2ff3c970
* src/tools: dfce0fbcdd..3cb5afca12
DEPS diff: 0a788fbaed..fa9574f1d1/DEPS

No update to Clang.

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

Change-Id: Ib07865347896762877e25558c6c5b6aca544a83c
Reviewed-on: https://webrtc-review.googlesource.com/c/120240
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@{#26429}
 2019-01-28 19:42:25 +00:00
Amit Hilbuch
bcd39d483d Creating Simulcast offer and answer in Peer Connection. 
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.

Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}
 2019-01-28 18:56:02 +00:00
Steve Anton
e76ca61238 Allow use of functions in absl/algorithms 
Bug: None
Change-Id: Id8311e6374228675cd34e413411611c77ed2d36d
Reviewed-on: https://webrtc-review.googlesource.com/c/119963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26427}
 2019-01-28 18:42:16 +00:00
Piotr (Peter) Slatala
48c5493393 Add 'UpdateAllocationLimits' in media transport. 
Bug: webrtc:9719
Change-Id: I90bd1d9858c259d7339420c574ad83d6fb18299c
Reviewed-on: https://webrtc-review.googlesource.com/c/118946
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26426}
 2019-01-28 18:20:47 +00:00
Niels Möller
435ea0a741 Add is_fec property to RtpPacketToSend 
Use instead of checking the packet's payload type and ssrc.

Bug: webrtc:7135
Change-Id: I272922a7879ef3e5e1344ce49044688572b9d942
Reviewed-on: https://webrtc-review.googlesource.com/c/120048
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26425}
 2019-01-28 15:43:21 +00:00
Artem Titov
a3ed451548 Add static factory method from FrameGenerator for FrameGeneratorCapturer. 
Add static factory method from FrameGenerator for FrameGeneratorCapturer
to be able to intercept generated frames in PC e2e test framework to
dump input video stream into file, if it was generated.

Bug: webrtc:10138
Change-Id: Iabecfaaef804111e0b19756cd676c1749757d9c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119947
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26424}
 2019-01-28 15:09:02 +00:00
Mirko Bonadei
37ec55e2bb [clang-tidy] Apply performance-faster-string-find fixes. 
This CL applies clang-tidy's performance-faster-string-find [1] on the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-faster-string-find.html

Bug: webrtc:10252
Change-Id: I4b8c0396836f3c325488e37d97037fa04742a5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/120047
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26423}
 2019-01-28 11:31:53 +00:00
Mirko Bonadei
190713c7cd Remove +api from internal DEPS files. 
This is redundant with [1].

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/DEPS?l=1424&rcl=914acd7589c3a31d8f99932b9c9a1917af2aa70f

Bug: webrtc:10244
No-Try: True
Change-Id: I447a9cb4187020d0ed74a2729b85d7924993cc70
Reviewed-on: https://webrtc-review.googlesource.com/c/119924
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26422}
 2019-01-28 11:17:00 +00:00
Åsa Persson
7d61352c7a Remove unused defines and methods in internal_defines.h 
Bug: none
Change-Id: Ia73dda32373fb367b6163f1157392c9d8077e4fc
Reviewed-on: https://webrtc-review.googlesource.com/c/116281
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26421}
 2019-01-28 10:31:40 +00:00
Mirko Bonadei
739baf097b [clang-tidy] Apply performance-for-range-copy fixes. 
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html

Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
 2019-01-28 09:53:50 +00:00
chromium-webrtc-autoroll
2d65fff16f Roll chromium_revision 53292b65a5..0a788fbaed (626349:626455) 
Change log: 53292b65a5..0a788fbaed
Full diff: 53292b65a5..0a788fbaed

Changed dependencies
* src/base: c6910d1a36..8889f1fcd9
* src/build: dede2d413f..a041d21740
* src/ios: 6cf0c3766a..f43b824a07
* src/testing: db0ccadd2f..6ed975ab13
* src/third_party: 6d904fb5e5..73ebf220db
* src/third_party/depot_tools: eb2767b2eb..bdb1123726
* src/third_party/googletest/src: 9518a57428..5ec7f0c4a1
* src/tools: 93e9054c12..dfce0fbcdd
DEPS diff: 53292b65a5..0a788fbaed/DEPS

No update to Clang.

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

Change-Id: I4804121ad89d64e7858def23c7f99ea5bc0ddc93
Reviewed-on: https://webrtc-review.googlesource.com/c/120142
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@{#26419}
 2019-01-28 07:33:22 +00:00
chromium-webrtc-autoroll
82709048d6 Roll chromium_revision 334d413a77..53292b65a5 (626249:626349) 
Change log: 334d413a77..53292b65a5
Full diff: 334d413a77..53292b65a5

Changed dependencies
* src/base: 952cb6b689..c6910d1a36
* src/build: 75934e6353..dede2d413f
* src/ios: 004450bb81..6cf0c3766a
* src/testing: 2e537d4ac6..db0ccadd2f
* src/third_party: e7a31775c7..6d904fb5e5
* src/third_party/depot_tools: db34d87aff..eb2767b2eb
* src/tools: 07542a3f6d..93e9054c12
DEPS diff: 334d413a77..53292b65a5/DEPS

No update to Clang.

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

Change-Id: I7ca6b7ba3885b35edfcc767456524f5a414431a4
Reviewed-on: https://webrtc-review.googlesource.com/c/120028
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@{#26418}
 2019-01-26 15:34:38 +00:00
Steve Anton
f380284035 (7) Rename files to snake_case: remove forwarding headers 
Bug: webrtc:10159
Change-Id: I2ba899e0283b953538c7941c8790213e36d7c70e
Reviewed-on: https://webrtc-review.googlesource.com/c/118561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26417}
 2019-01-26 00:33:46 +00:00
Piotr (Peter) Slatala
55b91b988f Only create no-op DTLS if media transport is used for both media and data 
Currently it's possible that no-op DTLS is created if media transport is only used for data channels.
Changing it so that no-op DTLS is only created when both media & data will flow through media transport.

Bug: webrtc:9719
Change-Id: I87f27fc778ea21b12f2904bad1452d893f66b541
Reviewed-on: https://webrtc-review.googlesource.com/c/119909
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26416}
 2019-01-26 00:04:22 +00:00
chromium-webrtc-autoroll
9058e076c0 Roll chromium_revision 3343618014..334d413a77 (626126:626249) 
Change log: 3343618014..334d413a77
Full diff: 3343618014..334d413a77

Changed dependencies
* src/base: 5bbe3caa9f..952cb6b689
* src/build: 838abed988..75934e6353
* src/ios: c4af087b33..004450bb81
* src/testing: e69083cab6..2e537d4ac6
* src/third_party: e2106465bd..e7a31775c7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1879ca54b9..514fe3e70d
* src/third_party/depot_tools: 60574b5f91..db34d87aff
* src/tools: e9cc7fad3f..07542a3f6d
DEPS diff: 3343618014..334d413a77/DEPS

No update to Clang.

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

Change-Id: I5d7dee7a4a316a1e37fd88f0b1d67015b0cccc84
Reviewed-on: https://webrtc-review.googlesource.com/c/119981
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@{#26415}
 2019-01-25 23:48:56 +00:00
Mirko Bonadei
d970807e0c Remove rtc_base/scoped_ref_ptr.h. 
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
 2019-01-25 20:29:58 +00:00
chromium-webrtc-autoroll
9444f3a8c8 Roll chromium_revision 6a5b2b19b1..3343618014 (626014:626126) 
Change log: 6a5b2b19b1..3343618014
Full diff: 6a5b2b19b1..3343618014

Changed dependencies
* src/base: 9015adf2da..5bbe3caa9f
* src/build: 018911f9a4..838abed988
* src/ios: 528045cd2a..c4af087b33
* src/testing: 5ee5c49371..e69083cab6
* src/third_party: bea3b73746..e2106465bd
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/8e8f250422..6c1b376e1d
* src/third_party/depot_tools: 80b9cf7dfd..60574b5f91
* src/tools: 91febde900..e9cc7fad3f
DEPS diff: 6a5b2b19b1..3343618014/DEPS

Clang version changed 351477:352138
Details: 6a5b2b19b1..3343618014/tools/clang/scripts/update.py

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

Change-Id: Ie9f02f7198f9f87e0f31b23bc976dee021f2bbb5
Reviewed-on: https://webrtc-review.googlesource.com/c/119961
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@{#26413}
 2019-01-25 19:38:16 +00:00
Jiawei Ou
d3a5aaa521 Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc 
Bug: webrtc:10241
Change-Id: I335718c81436502cc492c9142220cd023b7da80c
Reviewed-on: https://webrtc-review.googlesource.com/c/119860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#26412}
 2019-01-25 18:51:07 +00:00
Piotr (Peter) Slatala
63a176b34f Do not modify media transport config when falling back to RTP 
It is possible that media transport is re-set by the caller, but once
disabled it should stay disabled.

it's possible to fail this check the check in JsepTransportController::SetMediaTransportFactory in such case.

We should also change the caller to not invoke SetMediaTransportFactory
multiple times (with the same value), but I'll leave it as an excercise
to someone else :)

Bug: webrtc:9719
Change-Id: Ideea8a50d863edf4ef59e594a78c74bb9aba5aa7
Reviewed-on: https://webrtc-review.googlesource.com/c/119911
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26411}
 2019-01-25 18:19:17 +00:00
Bjorn Terelius
18f65dc20a Don't attempt to unwrap RTP timestamps for RTX stream. 
This fixes a bug where the event_log_visualizer hits a DCHECK when the RTP timestamp jumps.

TBR = kwiberg

Bug: webrtc:10170
Change-Id: I127a8e6165265d0726892a912f5bcdc33d98ced5
Reviewed-on: https://webrtc-review.googlesource.com/c/119664
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26410}
 2019-01-25 15:59:22 +00:00
Niels Möller
44b31d64ed Delete leftover method MaxConfiguredBitrateVideo and member remote_ssrc_ 
Bug: None
Change-Id: Ib2ed810fd02ce1d3d4b7c9f86f80668fb5242604
Reviewed-on: https://webrtc-review.googlesource.com/c/119954
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26409}
 2019-01-25 15:57:34 +00:00
Sebastian Jansson
0ef117e14c Improving robustness of stable bandwidth estimate test. 
It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.

Also fixing some minor test suite bug found during investigation.

Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
 2019-01-25 15:06:17 +00:00
Niels Möller
bebca61e5e Delete unused method SetSelectiveRetransmissions 
Bug: None
Change-Id: I5a59b5776fe537ec380629f9e5e9ac98c9e1214b
Reviewed-on: https://webrtc-review.googlesource.com/c/119920
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26407}
 2019-01-25 14:40:04 +00:00
Artem Titov
728b5a4033 Fix initialization to prevent SIGSEGV 
Bug: webrtc:10138
Change-Id: Ib299d2c5c08c07bbccf475b7e585cdd23830e238
Reviewed-on: https://webrtc-review.googlesource.com/c/119948
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26406}
 2019-01-25 14:38:02 +00:00