f6adac87b4
Add rtc event generic packet sent and received.
...
Bug: webrtc:9719
Change-Id: I2f692d9c1b33ac390975a9e695c7652cdc1b1e6e
Reviewed-on: https://webrtc-review.googlesource.com/c/121680
Commit-Queue: Peter Slatala <psla@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Bjorn Mellem <mellem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26616}
2019-02-08 19:13:57 +00:00
50930a6fb6
Roll chromium_revision 46a21d8d05..d60317bbda (630250:630357)
...
Change log: 46a21d8d05..d60317bbda46a21d8d05..d60317bbda4ddea1c782..1699ecbc53e3ed5e43c3..9a53be87ebc0d8777f9c..9d7fffdf63e056316509..369990ceaf448e819cf1..12208f82d8545f0d025e..610a4c6ce7b42269b000..6e7870157946a21d8d05..d60317bbdaTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I80e4758bc6800a9d54962887dd1caa46bdd3d912
Reviewed-on: https://webrtc-review.googlesource.com/c/122000
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@{#26615}
2019-02-08 17:52:31 +00:00
1d13b37b0c
Update LibvpxVp8Encoder to use EncodedImage::Allocate
...
Bug: webrtc:9378
Change-Id: I81bc1917e615e2982ba022a519bde9e5f55ab699
Reviewed-on: https://webrtc-review.googlesource.com/c/121840
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26614}
2019-02-08 17:01:36 +00:00
b7edf69e9a
Delete rtc::File, usage replaced with FileWrapper
...
Bug: webrtc:6463
Change-Id: Ia0767a2e6bbacc43e63c30ed3bd3edb10ff6e645
Reviewed-on: https://webrtc-review.googlesource.com/c/121943
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26613}
2019-02-08 16:23:53 +00:00
9f3aabb5ad
Delete obsolete class cricket::VideoCapturer
...
Bug: webrtc:6353
Change-Id: I220aca39319fd6562190f04bc97aa1fa9e523f31
Reviewed-on: https://webrtc-review.googlesource.com/c/119221
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26612}
2019-02-08 15:23:06 +00:00
494ff28573
Delete unused media constraints
...
Bug: webrtc:9239
Change-Id: I3a0a6b3f8d08bcc589e4f6490731fbe1598d0463
Reviewed-on: https://webrtc-review.googlesource.com/c/121820
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26611}
2019-02-08 14:45:00 +00:00
a8d48ab87b
Fix incorrect FPS measure when frame dropper kicks in
...
Bug: webrtc:10302
Change-Id: I4f8df7d41d8750e0810c2300fcd90b3eff7fb56d
Reviewed-on: https://webrtc-review.googlesource.com/c/121954
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26610}
2019-02-08 13:50:10 +00:00
bdfadd666e
Adds Stop methods to media streams in scenario framework.
...
Bug: webrtc:9510
Change-Id: If011e701496850dd67394052edd5a6d14a3998be
Reviewed-on: https://webrtc-review.googlesource.com/c/121951
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26609}
2019-02-08 13:21:20 +00:00
85eab49af4
Simplify peer connection smoke test to remove flakiness for now.
...
Bug: webrtc:10138
Change-Id: I81e9519eecab4195537524c542848c69d5b04100
Reviewed-on: https://webrtc-review.googlesource.com/c/121952
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26608}
2019-02-08 11:05:03 +00:00
3dd473b224
Refactor of RtpPacket constructor
...
Bug: None
Change-Id: I869d654cb28bc6d8291d77d6b0c45a68a4232a38
Reviewed-on: https://webrtc-review.googlesource.com/c/107887
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26607}
2019-02-08 10:59:02 +00:00
7ff164e6e1
Plumbing of feedback on request setting
...
Bug: webrtc:10263
Change-Id: I23c09e680d6381598e4172b76025ff84f33aa4de
Reviewed-on: https://webrtc-review.googlesource.com/c/121422
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26606}
2019-02-08 10:54:21 +00:00
5f6abcfbd2
Fix for RttBackoff when sending of packets with TWCC stops.
...
Bug: webrtc:10290
Change-Id: Ia825cbde070214e5ec9f5439246ea43f58c3c2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/121561
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26605}
2019-02-08 10:47:03 +00:00
dcba72b236
Resume rolling buildtools, now as chromium/src/buildtools
...
Based on https://chromium-review.googlesource.com/c/chromium/src/+/1450459/10/DEPS#294
Bug: chromium:927867
Change-Id: I6a69bb11ae0c2332b18c64ab630ea2c2c3285e59
Reviewed-on: https://webrtc-review.googlesource.com/c/121947
Commit-Queue: Oleh Prypin <oprypin@webrtc.org >
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26604}
2019-02-08 10:36:39 +00:00
b76b9ba8ce
Set WEBRTC_USE_H264 in common_config
...
Bug: none
Change-Id: I3dce8fdc409c88cdd771ea57eca3ea375e6e82c9
Reviewed-on: https://webrtc-review.googlesource.com/c/121948
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26603}
2019-02-08 10:17:04 +00:00
3f171df263
Add support for building iOS simulator code for iOS 11 and 12
...
No-Try: True
Bug: webrtc:10291
Change-Id: I7c9349bcacd2c26c28ec08a0a3b1b1b4c104ae12
Reviewed-on: https://webrtc-review.googlesource.com/c/121645
Reviewed-by: Oleh Prypin <oprypin@webrtc.org >
Commit-Queue: Artem Titarenko <artit@google.com >
Cr-Commit-Position: refs/heads/master@{#26602}
2019-02-08 10:07:49 +00:00
52e9e8d810
Remove now-unused iOS CI config files
...
Bug: webrtc:10253
Change-Id: Id952221470015bc2e986da4315de58b58b9256aa
Reviewed-on: https://webrtc-review.googlesource.com/c/120351
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Commit-Queue: Oleh Prypin <oprypin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26601}
2019-02-08 09:49:39 +00:00
51aa82d59b
Roll chromium_revision 6f2fb1192a..46a21d8d05 (630145:630250)
...
Change log: 6f2fb1192a..46a21d8d056f2fb1192a..46a21d8d05610e94b9c8..4ddea1c78204353fd2e5..c0d8777f9cf8bb316c14..e056316509cb9b45092c..448e819cf1https://chromium.googlesource.com/catapult.git/+log/38769c1f96..22b1689dde
* src/third_party/depot_tools: b9ae2ca9a5..545f0d025ea0c7fcb097..b42269b0006f2fb1192a..46a21d8d05TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I40dd2ac183ee75f45047421906c87ecd31e95cd1
Reviewed-on: https://webrtc-review.googlesource.com/c/121985
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@{#26600}
2019-02-08 09:36:47 +00:00
9f97c9a528
Add starting of VideoQualityAnalyzer in the e2e peer connection level test
...
Bug: webrtc:10138
Change-Id: Ic762543e21a5b55c7f15856fe752534b910dec8f
Reviewed-on: https://webrtc-review.googlesource.com/c/121941
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Peter Slatala <psla@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26599}
2019-02-08 09:35:27 +00:00
5963fddac2
Pass-by-reference instead of value to initWithNativeEncodedImage
...
Previously, the use of pass-by-value caused an issue in
ObjCVideoDecoder::Decode, where the EncodedImage was being copied upon
calling initWithNativeEncodedImage, which then created an NSData using
the copy's pointer; then the copy was destroyed, invalidating that
pointer.
Bug: webrtc:9378
Change-Id: Iac28b890c9902108ffc5ec54a607a99034159153
Reviewed-on: https://webrtc-review.googlesource.com/c/121922
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26598}
2019-02-08 08:56:23 +00:00
108f20fd45
Fix color space bug in wrapper of H264 decoder
...
Bug: none
Change-Id: I8309e5e3c177ae75712fa257e083ea2018a1f8e2
Reviewed-on: https://webrtc-review.googlesource.com/c/121760
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26597}
2019-02-08 08:47:53 +00:00
a8cb366f98
Add field trial for forced software decoder fallback.
...
Bug: none
Change-Id: Ibd8564fc243105274c2778ccfc99d38049c826b1
Reviewed-on: https://webrtc-review.googlesource.com/c/121720
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26596}
2019-02-08 07:09:35 +00:00
587c5d1617
Roll chromium_revision 34f99c21a3..6f2fb1192a (630023:630145)
...
Change log: 34f99c21a3..6f2fb1192a34f99c21a3..6f2fb1192ad029fef0da..610e94b9c8a721f3327e..e3ed5e43c36a1390bde0..04353fd2e54b50bc6568..f8bb316c14399bdf2333..cb9b45092c06d1040fab..b9ae2ca9a56d07835ab0..a0c7fcb09734f99c21a3..6f2fb1192aTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I9b86424917e0b510c4de9f9102722ffa76c313d8
Reviewed-on: https://webrtc-review.googlesource.com/c/121980
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@{#26595}
2019-02-08 04:07:21 +00:00
ec3b9ffdb0
Move audio-related MediaTransport interfaces to their own file and target
...
Bug: webrtc:9719
Change-Id: I8bef979e4073d51be7cb93d38ee0e2ae22baef0e
Reviewed-on: https://webrtc-review.googlesource.com/c/121942
Reviewed-by: Peter Slatala <psla@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26594}
2019-02-08 01:58:14 +00:00
e12778cb3a
Update VP9EncoderImpl to use EncodedImage::Allocate
...
Bug: webrtc:9378
Change-Id: I009138b4dc50c4ceb8f94fee6a958bbfa4d7e326
Reviewed-on: https://webrtc-review.googlesource.com/c/121771
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26593}
2019-02-07 21:26:43 +00:00
f9a5561de7
Roll chromium_revision ee5dfb2215..34f99c21a3 (629907:630023)
...
Change log: ee5dfb2215..34f99c21a3ee5dfb2215..34f99c21a31eb0493202..d029fef0daa4d3fbeca9..a721f3327ee02d2524ee..6a1390bde0d200b6e601..4b50bc65681242705f8e..399bdf2333cde3da57b9..ce4336c2ab9dd2ebe4c4..6d07835ab0ee5dfb2215..34f99c21a3TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,marpan@webrtc.org ,
BUG=None
Change-Id: I9623171b736d965610cc87d624e0e6c00e1f725d
Reviewed-on: https://webrtc-review.googlesource.com/c/121900
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@{#26592}
2019-02-07 20:23:43 +00:00
d7180cccc4
Also check the pending remote description when generating MIDs for legacy remote offers
...
Bug: webrtc:10296
Change-Id: Ia10299177175e57d3f494281310d6c91bed9ebdb
Reviewed-on: https://webrtc-review.googlesource.com/c/121860
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Amit Hilbuch <amithi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26591}
2019-02-07 20:22:38 +00:00
ce470aab51
Enabling Simulcast use via AddTransceiver.
...
This change removes the restriction on multiple send encodings when
calling AddTransceiver. If RIDs are not provided in the
simulcast scenario, they are auto-generated by the platform.
This effectively enables the use of spec-compliant simulcast.
Tests are also added to verify simulcast functionality.
Bug: webrtc:10075
Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/121389
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Amit Hilbuch <amithi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26590}
2019-02-07 17:29:59 +00:00
a6a273db11
Introduce PeerConnectionE2EQualityTestFixture implementation.
...
Introduce PeerConnectionE2EQualityTestFixture implementation with
example test.
Bug: webrtc:10138
Change-Id: Iec1d135f1b43863a3fa6f0723b579d2b7ff44807
Reviewed-on: https://webrtc-review.googlesource.com/c/120810
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Peter Slatala <psla@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Yves Gerey <yvesg@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26589}
2019-02-07 17:07:39 +00:00
c363a53587
Define RtpGenericFrameDescriptorExtension00
...
We are about to split RtpGenericFrameDescriptorExtension
into v00 and v01. Allow downstream projects to refer to
RtpGenericFrameDescriptorExtension00 now, so that we may later
delete references to RtpGenericFrameDescriptorExtension
without breaking their build.
Bug: webrtc:10214
Change-Id: I45528699bf7d8cc6c22c22a601f248cca2ba6c90
Reviewed-on: https://webrtc-review.googlesource.com/c/121769
Commit-Queue: Elad Alon <eladalon@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26588}
2019-02-07 16:50:18 +00:00
260a71d47c
Delete deprecated method PeerConnectionFactory::CreateVideoSource
...
Bug: webrtc:6353
Change-Id: Icb8847b234e7a844a4dff9ff44861f6967ac7b5b
Reviewed-on: https://webrtc-review.googlesource.com/c/118661
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26587}
2019-02-07 14:24:02 +00:00
59ab1cf081
Move ownership of RTPSenderVideo and RTPSenderAudio one level up
...
From RTPSender to RtpRtcpImpl. Makes RTPSender operate on packets
only, not frames.
Bug: webrtc:7135
Change-Id: Ia9a11456404c3b322d873d4f8fb828742296b26d
Reviewed-on: https://webrtc-review.googlesource.com/c/120044
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26586}
2019-02-07 13:31:48 +00:00
938dd9f1e8
Add owned data buffer to EncodedImage
...
Bug: webrtc:9378
Change-Id: I6a66b9301cbadf1d6517bf7a96028099970a20a3
Reviewed-on: https://webrtc-review.googlesource.com/c/117964
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26585}
2019-02-07 13:11:47 +00:00
e6f6a0cb8d
Add missing operator= and extra methods to the SamplesStatsCounter.
...
Add missing copy and move operator= and GetVariance and
GetStandardDeviation methods to the SamplesStatsCounter.
Change-Id: I02374aac23a00fdeefda16012311cd860bb4b1b5
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/121653
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26584}
2019-02-07 12:59:17 +00:00
710f3d3e44
Use task queue factory factory as parameter for TaskQueueTest
...
The new parameter simplify TaskQueueFactory lifetime and
allows insertion of per TestCase state
Bug: webrtc:10191
Change-Id: If4948df2756580957052b2b333b5c12cf4914d55
Reviewed-on: https://webrtc-review.googlesource.com/c/121648
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26583}
2019-02-07 11:00:17 +00:00
0041fe53dc
Roll chromium_revision 1a597bc4e4..ee5dfb2215 (629788:629907)
...
Change log: 1a597bc4e4..ee5dfb22151a597bc4e4..ee5dfb22151161fb86d1..1eb0493202e148b4cffc..a4d3fbeca93f7730ca8d..e02d2524eebbb6ca1af1..d200b6e601a8a90adc37..1242705f8ehttps://chromium.googlesource.com/catapult.git/+log/11e283fea2..38769c1f96
* src/third_party/depot_tools: 61d0c29253..06d1040faba02273b65b..9dd2ebe4c41a597bc4e4..ee5dfb2215TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I3344b447f43ba0e37104be3abea6a465efa57b2e
Reviewed-on: https://webrtc-review.googlesource.com/c/121740
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@{#26582}
2019-02-07 09:33:12 +00:00
cdab13d9a3
Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788)
...
Change log: c27b32b2fd..1a597bc4e4c27b32b2fd..1a597bc4e45b5c6b774e..1161fb86d1ae9b018ef6..e148b4cffc472ec5538f..3f7730ca8d73be5bfc61..bbb6ca1af1778cfca8c4..a8a90adc37https://boringssl.googlesource.com/boringssl.git/+log/1a51a5b4a6..70fe610556
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/78b1c61fff..11e283fea2
* src/third_party/depot_tools: 484d74fa21..61d0c292531e7a8f30c2..a6feefdfef662d7014c7..a02273b65bc27b32b2fd..1a597bc4e4c27b32b2fd..1a597bc4e4https://webrtc-review.googlesource.com/c/121702
Commit-Queue: Artem Titarenko <artit@webrtc.org >
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26581}
2019-02-07 08:39:57 +00:00
86c8ad954b
Pause rolling buildtools
...
This is done to un-break the roller script, so other deps can be rolled for now.
Example failure: https://ci.chromium.org/p/webrtc/builders/luci.webrtc.cron/Auto-roll%20-%20WebRTC%20DEPS/3040
chromium/buildtools is obsolete, to be replaced with chromium/src/buildtools when the subtree mirror is available.
No-Try: True
Bug: chromium:927867
Change-Id: I340b7d0a79dcb68d9c9e321c2da20476005625f2
Reviewed-on: https://webrtc-review.googlesource.com/c/121701
Commit-Queue: Artem Titarenko <artit@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26580}
2019-02-07 08:38:52 +00:00
ef288dd267
Reland: Remove dead code from stream_params.h
...
Bug: None
Change-Id: I257b09416b2328601beb3f3807b85d3065f63d43
Reviewed-on: https://webrtc-review.googlesource.com/c/121660
Reviewed-by: Amit Hilbuch <amithi@webrtc.org >
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26579}
2019-02-06 19:18:15 +00:00
e1dcce24e6
Remove HAVE_WEBRTC_VOICE.
...
Appears not used anymore.
Bug: none
Change-Id: Ic2238e6ad3d9917208bdb4a101f1ce254b1272ac
Reviewed-on: https://webrtc-review.googlesource.com/c/120963
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26578}
2019-02-06 18:39:45 +00:00
e7b9e6b17d
Move RtpSenderVideo tests to separate file.
...
Also refactor most of the RtpSender tests to not use the frame-level
method RTPSender::SendOutgoingData.
Bug: webrtc:7135
Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7
Reviewed-on: https://webrtc-review.googlesource.com/c/121461
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26577}
2019-02-06 18:00:39 +00:00
d70a1148ae
Delete MediaTransport method SetNetworkChangeCallback
...
Followup to https://webrtc-review.googlesource.com/c/src/+/121460 .
Bug: webrtc:9719
Change-Id: I6261c450379de22d916f4048fec89e5e67e300f8
Reviewed-on: https://webrtc-review.googlesource.com/c/121651
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Peter Slatala <psla@webrtc.org >
Commit-Queue: Peter Slatala <psla@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26576}
2019-02-06 17:46:19 +00:00
fe6e50f0af
Allow more than one registered network change callback in MediaTransport
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Adds methods AddNetworkChangeCallback and RemoveNetworkChangeCallback,
to replace SetNetworkChangeCallback. Needed because both VideoChannel and
VoiceChannel register such a callback.
Bug: webrtc:9719
Change-Id: Ic592b2d775d721a0f44ba0af88ed963bf02d73a3
Reviewed-on: https://webrtc-review.googlesource.com/c/121460
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Peter Slatala <psla@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26575}
2019-02-06 16:14:45 +00:00
3e6188868b
Roll chromium_revision 9d5d0c6635..c27b32b2fd (629245:629510)
...
Change log: 9d5d0c6635..c27b32b2fd9d5d0c6635..c27b32b2fd2d89ba7994..5b5c6b774eca78469929..ae9b018ef66da1dd4151..472ec5538fa723626311..73be5bfc6194003f6b57..778cfca8c4https://chromium.googlesource.com/catapult.git/+log/a308a9443c..78b1c61fff
* src/third_party/depot_tools: 367af22db5..484d74fa21919eb0324b..662d7014c79d5d0c6635..c27b32b2fd9d5d0c6635..c27b32b2fdhttps://webrtc-review.googlesource.com/c/121621
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Commit-Queue: Oleh Prypin <oprypin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26574}
2019-02-06 16:11:25 +00:00
7ca375c8ca
Implement encoder overshoot detector and rate adjuster.
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The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.
Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.
Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26573}
2019-02-06 15:54:11 +00:00
e98954c35e
Prevent updating state in the delay manager if the packet was reordered.
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Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.
This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.
There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.
Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26572}
2019-02-06 15:30:54 +00:00
9025bd5142
Separate AndroidVideoTrackSource::OnFrameCaptured from adaptation
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AndroidVideoTrackSource::OnFrameCaptured currently does adaptation
before passing frames on. We want to add video processing between
adaptation and delivering the frame to the rest WebRTC C++. This
CL prepares for that by splitting OnFrameCaptured() into a separate
adaptation step and delivery step.
Bug: webrtc:10247
Change-Id: Iab759bac7f3072d4552ece80d0b81fc3e634c64c
Reviewed-on: https://webrtc-review.googlesource.com/c/119952
Commit-Queue: Magnus Jedvert <magjed@webrtc.org >
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26571}
2019-02-06 14:33:59 +00:00
bb87f8a4a4
Delete unused/unsupported RetransmissionMode constants
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Configurability via SetSelectiveRetransmissions was deleted in
https://webrtc-review.googlesource.com/c/119920 .
Delete constants kRetransmitFECPackets and kRetransmitAllPackets,
which are never enabled in production code. Also move the declaration
of RetransmissionMode from rtp_rtcp_defines.h to rtp_sender_video.h,
to reduce visibility to applications.
Bug: None
Change-Id: I70dcf7532aa3415a2449d8d807c500c1f149bf6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120053
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26570}
2019-02-06 14:19:09 +00:00
0859142244
Add events processing to GetIceEvents.
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Bug: webrtc:10170
Change-Id: I91f58fe67552060ae50eea425637dd50479a9f17
Reviewed-on: https://webrtc-review.googlesource.com/c/121643
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26569}
2019-02-06 14:14:09 +00:00
4092d6fb05
Fix autoroller to skip entries without @revision in them
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The problem started with https://chromium-review.googlesource.com/1407489
No-Try: True
Bug: None
Change-Id: Iaa8b6f0101404890ac44cca559168279807c94c8
Reviewed-on: https://webrtc-review.googlesource.com/c/121620
Reviewed-by: Artem Titarenko <artit@webrtc.org >
Commit-Queue: Oleh Prypin <oprypin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26568}
2019-02-06 12:19:37 +00:00
6cfb403791
Fix test FrameGenerator to work with a single file source
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Bug: None
Change-Id: I645779379145c6c7b2d452dc1a15f9d9b97a3ee5
Reviewed-on: https://webrtc-review.googlesource.com/c/121641
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26567}
2019-02-06 11:59:03 +00:00
