Commit Graph

29094 Commits

Author SHA1 Message Date
1b3f4f9b45 Allow RtpPacketHistory encapsulator function to abort retransmit
Bug: webrtc:10633
Change-Id: I162b2c2f778e8e4c6f31307028db0c352ded2276
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142230
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28312}
2019-06-18 17:59:16 +00:00
4cbb4ef817 Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest
In https://chromium-review.googlesource.com/1650265 attributes like minSdkVersion were moved from AndroidManifest.xml to GN files. For WebRTC there were a few problems with that.
* We don't want to suppress UsesMinSdkAttributes lint but now there are these "invalid" manifest files that we can't exclude or discern. So disable this lint error.
  https://chromium-review.googlesource.com/c/chromium/src/+/1650265/14/build/android/AndroidManifest.xml
* We should specify the versions in GN files, so I did that here (by exactly copying the versions that are already in the targets' corresponding XML files), but we never want to get rid of them in the XML files. For now this information will just be duplicated (without any synchronicity check!) so there should be followup to this.

Change log: 6ae0f0cd4c..bf62d746a4
Full diff: 6ae0f0cd4c..bf62d746a4

Changed dependencies
* src/base: 9e5e9332df..e5a1d1f652
* src/build: 5a031748ec..2ef566e990
* src/buildtools: 6ae683be2f..6f3775ad6e
* src/buildtools/linux64: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: 2f5c817266..7f1a97d593
* src/testing: 1d4247de57..b1b36ff0d4
* src/third_party: 6f7cbf7c46..42e96c4074
* src/third_party/android_sdk/public: ki7EDQRAiZAUYlnTWR1XmI6cJTk65fJ-DNZUU1zrtS8C..xhyuoquVvBTcJelgRjMKZeoBVSQRjB7pLVJPt5C9saIC
* src/third_party/android_sdk/public: iIwhhDox5E-mHgwUhCz8JACWQCpUjdqt5KTY9VLugKQC..ppQ4TnqDvBHQ3lXx5KPq97egzF5X2FFyOrVHkGmiTMQC
* src/third_party/android_sdk/public: 4Y2Cb2LGzoc-qt-oIUIlhySotJaKeE3ELFedSVe6Uk8C..MSnxgXN7IurL-MQs1RrTkSFSb8Xd1UtZjLArI8Ty1FgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ed9fcf3f70..9e5dbd8b46
* src/tools: f58f33bca1..a9a4b8fc7b
DEPS diff: 6ae0f0cd4c..bf62d746a4/DEPS

No update to Clang.

Bug: chromium:891996
Change-Id: I773d6fa90e8083d934c84eecc1cb9d7d4496eca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142235
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28311}
2019-06-18 17:10:06 +00:00
b762b5b794 Fix potential signed overflow in IntervalBudget::set_target_rate_kbps
This methods multiplies an int with 500. This cl ensure that does not overflow.

BUG=none

Change-Id: I2c4c4c169baf3bbc2cac021e87e396c605cd1815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141860
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28310}
2019-06-18 15:42:54 +00:00
342f98b117 Fixes for flexfec crash in scenario tests.
Bug: webrtc:9510
Change-Id: I39bb4ed9afc4837f88f0db798495f34b685f4c24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142232
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28309}
2019-06-18 15:03:41 +00:00
58ee187554 Add support within PacedSender and pacer queue for owning rtp packets.
This CL builds on https://webrtc-review.googlesource.com/c/src/+/142165
It adds the parts within the paced sender that uses those send methods.
A follow-up will add the pre-pacer RTP sender parts. That CL will also
add proper integration testing. Here, I mostly add coverage for the new
send methods. When the old code-path is removed, all tests need to be
converted to exclusively use the owned path.

Bug: webrtc:10633
Change-Id: I870d9a2285f07a7b7b0ef6758aa310808f210f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142179
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28308}
2019-06-18 15:02:19 +00:00
b028c6a8ff Support __EMSCRIPTEN__ in rtc_base.
Bug: None
Change-Id: Ie0497e268b93ad15a164db85e8cefdb823d64a65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28307}
2019-06-18 12:44:00 +00:00
0c0c9693b6 Add/rewrite H264 VUI video signal type description.
The rewriter updates video signal parameters in VUI such that they
match to given webrtc::ColorSpace.

Bug: webrtc:10723
Change-Id: I8d0593e3cb727bfee7eb00e3f9ff0b41b93b78bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140881
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28306}
2019-06-18 11:41:43 +00:00
449888ef99 Cleanup of resources from removed remote bitrate estimate test framework.
Bug: webrtc:9883
Change-Id: Id18133a021b3a064b00f0f99b5f30ebb92e89067
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140945
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28305}
2019-06-18 10:22:01 +00:00
2c648f5fb7 Stop running 'bwe_simulations_tests'.
The test binary will be remove by
https://webrtc-review.googlesource.com/c/src/+/140945.

Bug: webrtc:9883
Change-Id: I4d5276c51796586615e089339fba1cf8b0010927
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142229
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28304}
2019-06-18 10:02:02 +00:00
e181440316 Fix documentation in BitrateAdjuster.
Bug: webrtc:10700
Change-Id: I743111b1d79d6236de1fd6c0021008f350bf0c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141407
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28303}
2019-06-18 09:20:50 +00:00
0f557fe1ba Removes unused dependency on RTP/RTCP from loss based controller.
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.

Bug: webrtc:10749
Change-Id: I73e9402136cc16902d177234d63369938f191e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142223
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28302}
2019-06-18 07:06:52 +00:00
61d8ee10b6 Roll chromium_revision 2ce8c83798..6ae0f0cd4c (669595:669703)
Change log: 2ce8c83798..6ae0f0cd4c
Full diff: 2ce8c83798..6ae0f0cd4c

Changed dependencies
* src/base: f40a167f21..9e5e9332df
* src/build: ce759c0e27..5a031748ec
* src/ios: bb5a15012a..2f5c817266
* src/testing: 3af51754fa..1d4247de57
* src/third_party: 0b22a680b4..6f7cbf7c46
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e4b470c91..ed9fcf3f70
* src/third_party/libFuzzer/src: e9b95bcfe2..b80b89cb68
* src/third_party/libjpeg_turbo: d78acdd58d..e1669e3707
* src/tools: 2fa1789ff8..f58f33bca1
DEPS diff: 2ce8c83798..6ae0f0cd4c/DEPS

No update to Clang.

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

Change-Id: Ib71758698483b0a1ac2ec51407dd99c545d79aff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142662
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@{#28301}
2019-06-17 18:02:08 +00:00
23026ee6c3 Adds SortedByReceiveTime to TransportPacketsFeedback.
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.

Bug: webrtc:10749
Change-Id: I4d8ec92643f1a0d4b5fdac2fbcb067766afc6548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142222
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28300}
2019-06-17 17:43:27 +00:00
873a7a98b3 Fix event_log_visualizer help text and default profile.
Bug: webrtc:10623
Change-Id: I628881039200bdeef5469d913c085ef50d3352c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141868
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28299}
2019-06-17 16:23:16 +00:00
9c771c2089 Add TrySendPacket() method to RTP modules.
This method will be called when PacedSender is using the new code path
that directly owns the packets to be sent.

It can be seen as combining a few features of the old code path:
* It checks if this is the correct RTP module and then sends, without
  the need for PacketRouter to poll multiple methods for SSRC etc first.
* It partly corresponds to TimeToSendPacket(), but RTX encapsulation
  now happens pre-pacer and FEC does not need to have a packet history,
  so most of that method is not used.
* It implements most of PrepareAndSendPacket(), such as updating header
  extensions, reporting stats and of course forwards to Transport. It
  now also handles the history a bit differently, since media packets
  will only be stored for potential retransmission post-pacer.

Bug: webrtc:10633
Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28298}
2019-06-17 15:16:00 +00:00
d9c900f068 Add Clone() to Vp8FrameBufferControllerFactory
Bug: None
Change-Id: Ie59c795361420695ca9cb363ec5b3035728306f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142227
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28297}
2019-06-17 13:16:38 +00:00
1dee91aec9 Roll chromium_revision d958b08217..2ce8c83798 (669491:669595)
Change log: d958b08217..2ce8c83798
Full diff: d958b08217..2ce8c83798

Changed dependencies
* src/base: 4c258d2e5e..f40a167f21
* src/build: e7d7afff6d..ce759c0e27
* src/buildtools: 9c81a611c6..6ae683be2f
* src/buildtools/linux64: git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9..git_revision:8c7f49102234f4f4b9349dcb258554675475e596
* src/buildtools/mac: git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9..git_revision:8c7f49102234f4f4b9349dcb258554675475e596
* src/buildtools/win: git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9..git_revision:8c7f49102234f4f4b9349dcb258554675475e596
* src/third_party: cc07ba42ee..0b22a680b4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ef2cd5094f..2e4b470c91
* src/tools: 25ff67a6ae..2fa1789ff8
DEPS diff: d958b08217..2ce8c83798/DEPS

Clang version changed 80fee25776c2fb61e74c1ecb1a523375c2500b69:49b965079b18f8aa485dd1156dd088d40b7ee465
Details: d958b08217..2ce8c83798/tools/clang/scripts/update.py

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

Change-Id: I4491a2c1cd11fd28f883587168aa3f33295f48e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142622
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@{#28296}
2019-06-17 09:43:58 +00:00
e93454a383 Removes AddAndRemoveOld from SendTimeHistory
Bug: webrtc:9883
Change-Id: Id9ae4b2485ba5ff201e77771774cfc83ccbda0d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28295}
2019-06-17 08:46:54 +00:00
7b99fa89ca Roll chromium_revision 447b80d261..d958b08217 (669391:669491)
Change log: 447b80d261..d958b08217
Full diff: 447b80d261..d958b08217

Changed dependencies
* src/build: 3872731d03..e7d7afff6d
* src/ios: 7f6d7d84ef..bb5a15012a
* src/testing: 39c7d7470d..3af51754fa
* src/third_party: 3d7c416b69..cc07ba42ee
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2c7abe0de5..ef2cd5094f
* src/tools: 8d6fe8cdf3..25ff67a6ae
DEPS diff: 447b80d261..d958b08217/DEPS

No update to Clang.

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

Change-Id: I94a1c6a6ed00f93cca8eaa6267167d58078dfa72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142302
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@{#28294}
2019-06-15 07:32:32 +00:00
2f42dab4eb Roll chromium_revision e5d3b43486..447b80d261 (669285:669391)
Change log: e5d3b43486..447b80d261
Full diff: e5d3b43486..447b80d261

Changed dependencies
* src/base: ad9a15e262..4c258d2e5e
* src/build: a37a0177bc..3872731d03
* src/ios: 0110a95735..7f6d7d84ef
* src/testing: a6cf18f9a2..39c7d7470d
* src/third_party: 210e510c6a..3d7c416b69
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e502dbcca0..2c7abe0de5
* src/third_party/libjpeg_turbo: 2a34770be9..d78acdd58d
* src/tools: 8ac5a88edc..8d6fe8cdf3
DEPS diff: e5d3b43486..447b80d261/DEPS

No update to Clang.

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

Change-Id: If1a6963a7366cd8b91630a705711859661e2b0cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142244
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@{#28293}
2019-06-14 22:57:38 +00:00
ef5e35dc48 Roll chromium_revision be90998a40..e5d3b43486 (669184:669285)
Change log: be90998a40..e5d3b43486
Full diff: be90998a40..e5d3b43486

Changed dependencies
* src/base: 041778a828..ad9a15e262
* src/build: 00fa9e48cf..a37a0177bc
* src/ios: 7b47ac5ede..0110a95735
* src/testing: 508c99017f..a6cf18f9a2
* src/third_party: 6309f126df..210e510c6a
* src/tools: fe44423560..8ac5a88edc
DEPS diff: be90998a40..e5d3b43486/DEPS

No update to Clang.

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

Change-Id: I3b2732a780089272046ba6d1056a63e5968442a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142240
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@{#28292}
2019-06-14 18:38:22 +00:00
e112bb84ef Adds support for abs send time extension in scenario tests.
Bug: webrtc:10742
Change-Id: I2fba97b23691b27c05dce17ca17c5cd13076616b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141871
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28291}
2019-06-14 16:26:27 +00:00
3d61ab12e6 Adds send time to ReceivedPacket struct.
Bug: webrtc:10742
Change-Id: I7e83d5ec2e23d1db38d02a0c883466ecdcd387c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141874
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28290}
2019-06-14 15:01:36 +00:00
8aba8fe851 Reland "Populate the GFD-00 for H264 and generic codecs."
This is a reland of d3c6f9ccffe88749fde8bc1320baa1fe2db15b6b

Original change's description:
> Populate the GFD-00 for H264 and generic codecs.
> 
> Bug: none
> Change-Id: I368eb38740314280db87aaf8e179e9bd0fc20c3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/103502
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28272}

Bug: none
Change-Id: Ic02590e5328783969d5480a8f413986ef7055f8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142168
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28289}
2019-06-14 14:47:06 +00:00
bdb6b39bed Let HardwareVideoEncoder cache result from codec.getOutputBuffers()
Bug: webrtc:9378
Change-Id: I27d6fa9780a0fbb4607ad4d05dabf4414fe6b091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142173
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28288}
2019-06-14 14:40:55 +00:00
5e953d7968 Insert startcodes for H264 single NALU packets.
Bug: none
Change-Id: I68ee9b84d75b67044d28a47fe740c79a2cd33517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142169
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28287}
2019-06-14 13:45:21 +00:00
856ca198e2 Delete unused method ReportBlockStats::AggregateAndStore
Bug: None
Change-Id: I5511593b1459b1501c08ac41aa9220ed54ed73ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142164
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28286}
2019-06-14 12:57:29 +00:00
0292e9703f Roll chromium_revision f165a6d5de..be90998a40 (669076:669184)
Change log: f165a6d5de..be90998a40
Full diff: f165a6d5de..be90998a40

Changed dependencies
* src/base: bf8f8970a8..041778a828
* src/build: be4d13ba95..00fa9e48cf
* src/buildtools: 09893e534d..9c81a611c6
* src/ios: 39a7f18090..7b47ac5ede
* src/testing: f9ca2dafb6..508c99017f
* src/third_party: 5fbfb27850..6309f126df
* src/tools: 27a538e9e1..fe44423560
DEPS diff: f165a6d5de..be90998a40/DEPS

No update to Clang.

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

Change-Id: I7068b8d0dcf78e6f59dce4896474e5c271d81d11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142180
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@{#28285}
2019-06-14 12:49:29 +00:00
607a6f1c55 Moves conversion to ReceivedPacket from RtpPacketReceived to Call.
This moves the conversion from RtpPacketReceived to ReceivedPacket to
Call rather than RtpTransportController. This prepares for reusing the
struct for receive side network state estimation.

Bug: webrtc:10742
Change-Id: I9581438bc912ef4bb635a5d9a6dea488cf871d48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141872
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28284}
2019-06-14 12:19:49 +00:00
98cbb22911 Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream.
Bug: none
Change-Id: Ie90c9d6fd415678affcc4acefdd045334d83d8b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140864
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28283}
2019-06-14 11:39:42 +00:00
7f2a67f8ce Cleanup names and extra checks for errors in PC test framework
Bug: webrtc:10138
Change-Id: I5585f30e941cf9914028a0040388a02778ab46c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141672
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28282}
2019-06-14 11:31:52 +00:00
12d64deb6c Remove sequence_number from RtpPacketInfo.
This change removes sequence_number from RtpPacketInfo since it's currently not used.

Bug: webrtc:10668
Change-Id: I9b45c7476457df1d18173f37c421374108678931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141873
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28281}
2019-06-14 11:21:42 +00:00
ad82e8e1a6 Fix: restore disabling PC smoke test on iOS
Bug: webrtc:10138
Change-Id: I988cf96a60b7de6d304020135f53f90e8536feb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142161
Reviewed-by: Oleksandr Iakovenko <iakovenko@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28280}
2019-06-14 10:31:39 +00:00
19a1d50ccf Refactor WavWriter to use FileWrapper rather than PlatformFile
Bug: webrtc:6463
Change-Id: I4c80995481ed7d5c1079450d04ed7958fa137e84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141662
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28279}
2019-06-14 10:18:28 +00:00
04e129ab1d Revert "Populate the GFD-00 for H264 and generic codecs."
This reverts commit d3c6f9ccffe88749fde8bc1320baa1fe2db15b6b.

Reason for revert: Break downstream perf tests.

Original change's description:
> Populate the GFD-00 for H264 and generic codecs.
> 
> Bug: none
> Change-Id: I368eb38740314280db87aaf8e179e9bd0fc20c3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/103502
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28272}

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

Change-Id: I8582099dfca3a2acbf434214a3cf29572d7ad647
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28278}
2019-06-14 09:04:21 +00:00
1a285e081d Roll chromium_revision 96eca2d491..f165a6d5de (668951:669076)
Change log: 96eca2d491..f165a6d5de
Full diff: 96eca2d491..f165a6d5de

Changed dependencies
* src/base: 0a5b5a0b3c..bf8f8970a8
* src/build: 6b14af32fd..be4d13ba95
* src/ios: 0d988ccf0e..39a7f18090
* src/testing: 84fae89f8b..f9ca2dafb6
* src/third_party: dc5f8d5261..5fbfb27850
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/727d7ca273..e502dbcca0
* src/third_party/depot_tools: a74bd78e9c..bc23ca13f1
* src/tools: 3c454539c9..27a538e9e1
DEPS diff: 96eca2d491..f165a6d5de/DEPS

No update to Clang.

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

Change-Id: I648efe89357646b85353e655ff6d00fa1fdd3fe4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142100
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@{#28277}
2019-06-14 01:40:37 +00:00
a9a2a753e1 Roll chromium_revision 22f062d604..96eca2d491 (668845:668951)
Change log: 22f062d604..96eca2d491
Full diff: 22f062d604..96eca2d491

Changed dependencies
* src/base: 6f056dd3eb..0a5b5a0b3c
* src/build: 377edc00fe..6b14af32fd
* src/ios: da70dae394..0d988ccf0e
* src/testing: 1b0ab755f1..84fae89f8b
* src/third_party: 1741116d91..dc5f8d5261
* src/third_party/depot_tools: 374a128977..a74bd78e9c
* src/tools: 20ba638055..3c454539c9
DEPS diff: 22f062d604..96eca2d491/DEPS

No update to Clang.

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

Change-Id: Ifb08ed476e3c8668560e98c389f55b3c3db4ba5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142040
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@{#28276}
2019-06-13 22:19:25 +00:00
c5effc2453 Remove DataContentDescription shim
Bug: webrtc:10597
Change-Id: Id0cbb78846d2b248bc2ab650eb7c06b50bc825bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28275}
2019-06-13 18:33:40 +00:00
ef3fd9c8ad Add support for simulcast with Vp8 from caller into PC level quality tests.
Add support of negotiating simulcast offer/answer. Also fix some minor
issues around to make it finally work.

Bug: webrtc:10138
Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28274}
2019-06-13 17:27:09 +00:00
6751260241 Roll chromium_revision b08bd9b643..22f062d604 (668716:668845)
Change log: b08bd9b643..22f062d604
Full diff: b08bd9b643..22f062d604

Changed dependencies
* src/base: 68f52403e3..6f056dd3eb
* src/build: afe9cbd54d..377edc00fe
* src/buildtools: 5b97b40654..09893e534d
* src/buildtools/third_party/libc++/trunk: 78822a6853..5938e0582b
* src/ios: 0e29396fe5..da70dae394
* src/third_party: e832ae1894..1741116d91
* src/third_party/android_deps/libs/com_google_ar_core: version:1.8.0-cr0..4_5y1Cw_L1MHu3UedmkavqbZ7H7sYPBTdpcAOJQvlXkC
* src/third_party/freetype/src: d1b16325e2..e7ac9288ac
* src/tools: 4ae0c7f02a..20ba638055
DEPS diff: b08bd9b643..22f062d604/DEPS

No update to Clang.

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

Change-Id: I655b638d21314f80ac886c9786d0766f3e12d544
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141960
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@{#28273}
2019-06-13 16:42:12 +00:00
d3c6f9ccff Populate the GFD-00 for H264 and generic codecs.
Bug: none
Change-Id: I368eb38740314280db87aaf8e179e9bd0fc20c3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/103502
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28272}
2019-06-13 16:40:32 +00:00
9e25f74faa Update visibility for JNI targets in sdk/android.
Bug: webrtc:9048
Change-Id: I16f77f3bf77e3fdfa8fd9792a6ab13aa872c32f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141869
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28271}
2019-06-13 16:05:23 +00:00
5894b6aad4 Add kPayloadTypeGeneric to CallTest and use it in VideoQualityTest.
- Add CallTest::kPayloadTypeGeneric
- Allow for unrecognized codec names in VideoQualityTest.
  Generic packetization is implicitly enabled for these codecs.

Tested: autoninja -C out/Debug && out/Debug/video_loopback
Bug: webrtc:10738
Change-Id: I57001be997db2f0eed9197eb40801b5ad936d222
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141864
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28270}
2019-06-13 15:45:00 +00:00
5740afa0a4 Removes SimulatedTimeClient
Bug: webrtc:9883
Change-Id: Id6e760b37360e7dafc67ded99e06128be20797d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141417
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28269}
2019-06-13 15:37:10 +00:00
6fd67f086c Pass java EncodedImage over jni to VideoEncoderWrapper::OnEncodedFrame
Preparation for adding a release() method on java's EncodedImage, and
call that from C++.

Bug: webrtc:9378
Change-Id: I301f64b16684c535f45a3fc9cd9ae1543df59d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141861
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28268}
2019-06-13 15:19:10 +00:00
f3f57700a8 Using full scenario test client for loss based control test.
Bug: webrtc:9883
Change-Id: I7c3b2561ddba846e4cdde05e1067679ada14ad80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141405
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28267}
2019-06-13 14:44:09 +00:00
4284828887 Remove deprecated version of RtpPacket::SetPadding that used to randomize padding
was deprecated in
https://webrtc-review.googlesource.com/c/src/+/103983

Bug: None
Change-Id: I617b7b5112446deaa9be983978cabdb247638266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28266}
2019-06-13 14:38:38 +00:00
5a8f860a01 Prepare for deletion of the NO_MAIN_THREAD_WRAPPING preprocessor define
This is a partial reland of
https://webrtc-review.googlesource.com/c/src/+/39680,
including only the (hopefully) non-problematic parts of it, but
postponing actual deletion of automatic thread wrapping.

Bug: webrtc:9714
Change-Id: I9b79dd073f0e945cbb62f3b54cff05eaaea9b06c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141664
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28265}
2019-06-13 13:51:17 +00:00
50dd80b96f Remove data channel only .so-file.
Bug: webrtc:10733
Change-Id: Ia08b9a03e41442f1b0407df575fc9f7d0a1bc86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141415
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28264}
2019-06-13 13:37:55 +00:00
3c396e52da Add injectable video encoder and decoder to video quality test.
Bug: webrtc:10738
Change-Id: Ia5180cf0252ecd1c58a2080e3954fcb886b066e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141667
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28263}
2019-06-13 13:03:05 +00:00