Commit Graph

26810 Commits

Author SHA1 Message Date
e0b9355aa8 Move enum VideoType out of common_types.h
New location is common_video/libyuv/include/webrtc_libyuv.h.

Bug: webrtc:5876
Change-Id: Ied439a83417008a086bd496a8d13042398ff1e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131330
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27478}
2019-04-08 09:47:54 +00:00
0d32a737dc Fix naming in NetworkEmulationManager: endpoint_controller -> endpoint_container
Bug: webrtc:10138
Change-Id: If5d6a9e6b25619278d32477e6c44c8fd5ad0faf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131331
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27477}
2019-04-08 09:35:39 +00:00
5a0665bea4 Make UDP receive buffer size configurable via field trial
Bug: chromium:939340
Change-Id: I2ab18554d12a1e9c62f5d3d8f8237cc4d0a1a78c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131395
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27476}
2019-04-08 09:31:39 +00:00
d1c6085cb3 Added FrameDecryptorInterface::Result constructor and IsOk() member function.
Bug: webrtc:10512
Change-Id: I48bdaad57739382b5c1040d94f4e3657e2054e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131364
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27475}
2019-04-08 07:04:07 +00:00
f948eb66aa Implement DefaultAudioQualityAnalyzer.
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.

When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms

Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
2019-04-07 14:32:33 +00:00
0b2bf9590f Roll chromium_revision 58ec016f06..baccefbc73 (648432:648532)
Change log: 58ec016f06..baccefbc73
Full diff: 58ec016f06..baccefbc73

Changed dependencies
* src/base: f41d59faf0..7f42cc431e
* src/build: 25794eb5ee..3eaa797fa2
* src/ios: be52df3518..d927186141
* src/third_party: f26e86be04..bbcbf51c30
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/bf0179e27c..979fc35bbc
* src/third_party/depot_tools: 9198ef8ede..0b62ed79ed
* src/tools: 1920aa9d40..d151b62130
DEPS diff: 58ec016f06..baccefbc73/DEPS

No update to Clang.

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

Change-Id: I6a55b450a2e8b153a647ddb7bf494e7d51eafb3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131600
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@{#27473}
2019-04-07 09:16:49 +00:00
a5b8220f2e Roll chromium_revision 0a611f37b8..58ec016f06 (648329:648432)
Change log: 0a611f37b8..58ec016f06
Full diff: 0a611f37b8..58ec016f06

Changed dependencies
* src/base: 1d63e15e13..f41d59faf0
* src/build: 23a5e22ba7..25794eb5ee
* src/ios: 1b45cc0d32..be52df3518
* src/testing: 8a96030944..a428665e17
* src/third_party: 5c95ab592b..f26e86be04
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/27980609ac..bf0179e27c
* src/third_party/depot_tools: dbc721d65f..9198ef8ede
* src/tools: f63001b39e..1920aa9d40
DEPS diff: 0a611f37b8..58ec016f06/DEPS

No update to Clang.

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

Change-Id: Iacdbe2c09deb55f99f1eae1e57c1f65afe6fbcba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131500
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@{#27472}
2019-04-06 01:29:38 +00:00
0694d1fce5 Roll chromium_revision 37663bcca7..0a611f37b8 (648215:648329)
Change log: 37663bcca7..0a611f37b8
Full diff: 37663bcca7..0a611f37b8

Changed dependencies
* src/base: 9ddc023f80..1d63e15e13
* src/ios: fe51fcc2c1..1b45cc0d32
* src/testing: 8b5edd181c..8a96030944
* src/third_party: a5a1eb9b53..5c95ab592b
* src/third_party/android_deps/libs/com_google_ar_core: version:1.6.0-cr0..version:1.8.0-cr0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9057b413d4..27980609ac
* src/tools: d6f4e3aa56..f63001b39e
DEPS diff: 37663bcca7..0a611f37b8/DEPS

No update to Clang.

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

Change-Id: I4d66339337e19d7827f26c19bab96d4f52ec85ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131442
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@{#27471}
2019-04-05 22:14:43 +00:00
72e977135c Add Result FrameDecryptorInterface::Decrypt
This change adds FrameDecryptorInterface::Result to the FrameDecryptorInterface
API. Result contains a Status and bytes_written. This removes requiring out
parameters from the API and provides a simpler status return code for the
function. This is in response to comments suggested here:
https://webrtc-review.googlesource.com/c/src/+/131358

int FrameDecryptorInterface::Decrypt() will be removed in a follow up CL.

Bug: webrtc:10512
Change-Id: I47f19f154d1d8430acd6e4a6f433ab24c455fd51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131362
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27470}
2019-04-05 21:07:16 +00:00
2a3cf05af7 Roll chromium_revision f91f825874..37663bcca7 (648096:648215)
Change log: f91f825874..37663bcca7
Full diff: f91f825874..37663bcca7

Changed dependencies
* src/base: fb54341f11..9ddc023f80
* src/build: e4948d79f2..23a5e22ba7
* src/ios: 8e699376f3..fe51fcc2c1
* src/testing: c562c291be..8b5edd181c
* src/third_party: af977ed4ee..a5a1eb9b53
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/810aaa1e18..9057b413d4
* src/tools: 973c47f044..d6f4e3aa56
DEPS diff: f91f825874..37663bcca7/DEPS

Clang version changed 356356:357692
Details: f91f825874..37663bcca7/tools/clang/scripts/update.py

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

Change-Id: Ia2871f209fe49c1ca016cc3391121ecbbaaa465a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131421
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@{#27469}
2019-04-05 17:32:37 +00:00
ebd94f6df1 Using simulated time for GoogCC tests.
Bug: webrtc:10365
Change-Id: I482e544f1585fdb54dc49740ba81870104dd58a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130509
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27468}
2019-04-05 15:58:29 +00:00
8f32b6c18c AEC3: Enable usage of external delay estimator
This change makes it possible to disable AEC3's render delay
controller and delay estimator, and instead rely on an external
delay estimator. The delay is communicated via SetAudioBufferDelay.

When the feature is enabled, no echo removal will be performed
until the first delay is provided.

The delay is

Bug: b/130016532
Change-Id: I16643109d78d770ff1d2713cf247b0b9cce1bc1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131327
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27467}
2019-04-05 15:34:39 +00:00
a553c72659 Tune VideoCodecTestLibvpx.TemporalLayersVP8 thresholds.
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.

Bug: none
Change-Id: I070a62ce725b0a79e5cb5c4679ab643de50c46c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131334
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27466}
2019-04-05 14:43:40 +00:00
363fb7e050 Running scenario quality unit tests in simulated time.
This is to avoid inconsistent/flaky behavior on mobile bots.

Bug: webrtc:10365
Change-Id: I52ab4f9ef92b10329c1eac502adfcf2886058114
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27465}
2019-04-05 14:35:40 +00:00
7b7485b796 Remove TaskQueue constructor that uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: I9547fb7110222ce3a3c2323ae2a004024eab911e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130471
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27464}
2019-04-05 13:34:26 +00:00
59e875ce18 Tune VideoCodecTestLibvpx.MultiresVP8 thresholds.
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.

Bug: none
Change-Id: Ifa661eea15a0d52f4760f4aac9294074faab757f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27463}
2019-04-05 13:15:34 +00:00
c46a999342 Reduce flakiness of repeating task test.
Bug: webrtc:9883
Change-Id: I9027de52dc6e3e20bbd7b5b977116b3be9077941
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131324
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27462}
2019-04-05 12:58:04 +00:00
ae2213b38d Delete compatibility alias webrtc::kI420
Bug: webrtc:7385, webrtc:10198
Change-Id: Ib18b6fc45f1bac7f4145d4fca384978a85b6c85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130481
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27461}
2019-04-05 11:47:38 +00:00
ff39312958 Add ability to have multiple connected remote endpoints
Bug: webrtc:10138
Change-Id: Ic305c2f247588d75b6ced17052ba12d937d1a056
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128864
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27460}
2019-04-05 10:27:14 +00:00
5684af5d63 VideoSendStream::Stats::total_encode_time_ms added.
This is a standard stat:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime

This is collected by SendStatisticsProxy. A follow-up CL will plumb
this to the RTCStatsCollector.

Bug: webrtc:10448
Change-Id: I236afa5576edc26afd54bd166f7faaf7e38e7c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130517
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27459}
2019-04-05 10:16:14 +00:00
a556448138 Don't recreate the VideoReceiveStream on SetFrameDecryptor in the MediaEngine.
This change introduces new logic to allow the injection of the FrameDecryptor
into an arbitrary already running VideoReceiveStream without resetting it. It
does this by taking advantage of the BufferedFrameDecryptor which will
forcefully be created regardless of whether a FrameDecryptor is passed in
during construction of the VideoReceiver if the
crypto_option.require_frame_encryption is true. By allowing the
BufferedFrameDecryptor to swap out which FrameDecryptor it uses this allows the
Receiver to switch decryptors without resetting the stream.

This is intended to mostly be used when you set your FrameDecryptor at a point
post creation for the first time.

Bug: webrtc:10416
Change-Id: If656b2acc447e2e77537cfa394729e5c3a8b660a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130361
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27458}
2019-04-05 07:58:05 +00:00
144575b65a Roll chromium_revision e05071635b..f91f825874 (647992:648096)
Change log: e05071635b..f91f825874
Full diff: e05071635b..f91f825874

Changed dependencies
* src/base: 6a69ac92e1..fb54341f11
* src/build: 372b912a6d..e4948d79f2
* src/ios: a101508a26..8e699376f3
* src/testing: 2c43c44509..c562c291be
* src/third_party: 76e8ba9855..af977ed4ee
* src/third_party/depot_tools: 9f74913e51..dbc721d65f
* src/tools: 7fbfd73192..973c47f044
DEPS diff: e05071635b..f91f825874/DEPS

No update to Clang.

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

Change-Id: I03b78a15943c3a4f92de2f9fe484fc39cd60778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131361
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@{#27457}
2019-04-05 07:18:44 +00:00
4e95e17a66 Roll chromium_revision b95224d1f4..e05071635b (647872:647992)
Change log: b95224d1f4..e05071635b
Full diff: b95224d1f4..e05071635b

Changed dependencies
* src/base: a90642899f..6a69ac92e1
* src/build: 089a33ed0f..372b912a6d
* src/ios: 07a8f68d15..a101508a26
* src/third_party: 57a96905da..76e8ba9855
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/144122958b..810aaa1e18
* src/third_party/icu: b10cc9f714..69c72a6dfe
* src/tools: 8491393d76..7fbfd73192
DEPS diff: b95224d1f4..e05071635b/DEPS

No update to Clang.

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

Change-Id: I4066fa5c905d60e9254637812bdb3295d90c7fdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131350
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@{#27456}
2019-04-05 01:02:21 +00:00
0ae222a62f Roll chromium_revision 4aff3b8e0f..b95224d1f4 (647765:647872)
Change log: 4aff3b8e0f..b95224d1f4
Full diff: 4aff3b8e0f..b95224d1f4

Changed dependencies
* src/base: 178202a54c..a90642899f
* src/build: 96d2ee3e71..089a33ed0f
* src/ios: cfa6ed29d0..07a8f68d15
* src/testing: cac71cb59f..2c43c44509
* src/third_party: 68b93d0ceb..57a96905da
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/63e0762df0..144122958b
* src/third_party/depot_tools: b8268cad11..9f74913e51
* src/third_party/libvpx/source/libvpx: ecae7f8f81..4117995a8e
* src/tools: 9ef99fc9b2..8491393d76
DEPS diff: 4aff3b8e0f..b95224d1f4/DEPS

No update to Clang.

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

Change-Id: Ic33bab3541a2c0ea590e545e7590cae440f73d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131347
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@{#27455}
2019-04-04 20:46:15 +00:00
1c747f5717 Preparing VideoReceiveStream for move to TaskQueue.
Extracting the work that's thread dependent from the work that will
also be done when using task queue.

Bug: webrtc:10365
Change-Id: I648796fe016c966c731c9b7f85d2a871c1f2a349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131241
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27454}
2019-04-04 17:01:42 +00:00
f75d458951 Roll chromium_revision e839e72818..4aff3b8e0f (647650:647765)
Change log: e839e72818..4aff3b8e0f
Full diff: e839e72818..4aff3b8e0f

Changed dependencies
* src/base: d04d0d4279..178202a54c
* src/build: 6fdec9fe12..96d2ee3e71
* src/ios: db67ecd5eb..cfa6ed29d0
* src/third_party: 7b588e18a3..68b93d0ceb
* src/tools: 75c78ba595..9ef99fc9b2
DEPS diff: e839e72818..4aff3b8e0f/DEPS

Clang version changed 357569:356356
Details: e839e72818..4aff3b8e0f/tools/clang/scripts/update.py

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

Change-Id: I792578151ff82d351b5030c591e4657e1755a749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131343
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@{#27453}
2019-04-04 16:32:49 +00:00
50b8c399c9 Generalize the C-language Opus interface.
Switch to explicit channel mappings (RFC 7845) when creating
multi-stream Opus en/de-coders. The responsibility of setting up the
channel mappings will shift from WebRTC to the WebRTC user.

See https://webrtc-review.googlesource.com/c/src/+/121764 for the
current vision. See also the first child CL
https://webrtc-review.googlesource.com/c/src/+/129768
that sets up the Decoder to use this code.

Bug: webrtc:8649
Change-Id: I55959a293d54bb4c982eff68ec107c5ef8666c5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129767
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27452}
2019-04-04 14:06:44 +00:00
21f6fd79ad Add preemptive rate and preferred buffer size plots to event log visualizer.
Bug: None
Change-Id: I185f1cd9645a850ba42cb122373d73613d118ac1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131128
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27451}
2019-04-04 13:50:29 +00:00
ec51ce0ce1 AEC3: Remove unused config parameters
This change removes the following unused parameters from the AEC3
configuration:
- render_pre_window_size_init
- render_post_window_size_init
- nonlinear_hold
- nonlinear_release

Bug: webrtc:8671
Change-Id: I8f7a3d350387cd8ada4d507c3a9fab43b7813f5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131321
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27450}
2019-04-04 12:36:48 +00:00
6c371ca700 Add OnLossNotification() to VideoEncoder and Vp8FrameBufferController
Bug: webrtc:10501
Change-Id: I33e8bfcf16cf24aadcfdf214d7d9bcd495bf9348
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131021
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27449}
2019-04-04 10:57:02 +00:00
f88aa97c52 Introduce RtpSequenceNumberMap
The video encoder should be informed of incoming LossNotification
RTCP messages. Since it is unaware of RTP sequence numbers,
or anything else RTP-related, the sequence numbers mentioned
in the RTCP message must first be mapped to timestamps,
since those are meaningful to the encoder.

This CL introduces RtpSequenceNumberMap, which maps RTP sequence
numbers to timestamps, while providing:
1. Capping the number of entries.
2. Wrap-around handling.

RtpSequenceNumberMap also remembers which packets were first
and/or last in the frame.

Later CLs will wire this up.

Bug: webrtc:10501
Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27448}
2019-04-04 10:50:12 +00:00
086b9073d4 Update codecs/h264 owners.
- Removed hbos@webrtc.org
- Added ssilkin@webrtc.org

Bug: none
Change-Id: I520be843cbb28652393cb0e0e05d7e0c1786e562
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131125
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27447}
2019-04-04 10:21:32 +00:00
1770ddb7a4 Remove obsolete TODO in default_encoded_image_data_injector.h
Bug: webrtc:10138
Change-Id: I401dea607feb2f1ea967dd2473bf4f0ba7a7d43e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27446}
2019-04-04 09:31:16 +00:00
2579047cc0 Roll chromium_revision 8524e2aded..e839e72818 (647205:647650)
Change log: 8524e2aded..e839e72818
Full diff: 8524e2aded..e839e72818

Changed dependencies
* src/base: 4d7d3409f7..d04d0d4279
* src/build: c2f2e5ab0f..6fdec9fe12
* src/buildtools: 459baaf66b..218cb3d12e
* src/buildtools/linux64: git_revision:0790d3043387c762a6bacb1ae0a9ebe883188ab2..git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b
* src/buildtools/mac: git_revision:0790d3043387c762a6bacb1ae0a9ebe883188ab2..git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b
* src/buildtools/third_party/libc++/trunk: 955113db37..fbddc46986
* src/buildtools/win: git_revision:0790d3043387c762a6bacb1ae0a9ebe883188ab2..git_revision:64b846c96daeb3eaf08e26d8a84d8451c6cb712b
* src/ios: b5be7e7906..db67ecd5eb
* src/testing: 946f832584..cac71cb59f
* src/third_party: 8ae85677bd..7b588e18a3
* src/third_party/android_deps/libs/android_arch_core_common: version:1.0.0-cr0..version:1.1.1-cr0
* src/third_party/android_deps/libs/android_arch_lifecycle_common: version:1.0.0-cr0..version:1.1.1-cr0
* src/third_party/android_deps/libs/android_arch_lifecycle_runtime: version:1.0.0-cr0..version:1.1.1-cr0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e1fc0b0b3d..63e0762df0
* src/third_party/depot_tools: 865445eb8a..b8268cad11
* src/tools: f05d4a5db9..75c78ba595
Added dependency
* src/third_party/android_deps/libs/android_arch_lifecycle_common_java8
DEPS diff: 8524e2aded..e839e72818/DEPS

No update to Clang.

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

Change-Id: If7c3d904c8b2394f820ee5c80f2931722d00f020
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131227
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@{#27445}
2019-04-04 09:14:26 +00:00
4bac79ece2 Add SetJitterBufferMinimumDelay method to RtpReceiverInterface.
This change is required to allow modification of Jitter Buffer delay
in javascript via Origin Trial Experiment.
Link to experiment description:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc

Bug: webrtc:10287
Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27444}
2019-04-04 09:00:16 +00:00
ada9b89b99 Added more refined benchmarking code for audioproc_f
This CL extends, and partly corrects, the benchmarking
code in audioproc_f to provide statistics for the API
call durations in audioproc_f

Bug: chromium:939791
Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27443}
2019-04-04 08:37:16 +00:00
1c1b1ea137 Allow setting ALR values for screen content again
When ALR was made default-on we removed the ability to use field trials
to configure alternative ALR detector values. This CL just restores
the ability to force them, defaults are unaffected.

Bug: webrtc:10509
Change-Id: Ibc09e27f1f7b72513de1482d280683802e962497
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131145
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27442}
2019-04-03 19:28:19 +00:00
5982d008a8 Stop always predicting from last keyframe in the 3TL VP8 case.
Bug: webrtc:10314
Change-Id: I510c84cce0ec05ad8ef977d57cba9585aabc0538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131141
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27441}
2019-04-03 15:16:30 +00:00
ade945d834 Add ability to specify encoder bitrate multiplier in PC level tests
Bug: webrtc:10138
Change-Id: I40b42e83ccec7b08226606d2770f3afa80e3fcc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130241
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27440}
2019-04-03 12:09:42 +00:00
fd720b2406 Switch to SendTask instead of manually waiting for event.
Bug: webrtc:10349
Change-Id: I128856d2baf221d67e957ce0614b075ecef3c5fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131140
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27439}
2019-04-03 12:03:14 +00:00
0682850c77 Roll chromium_revision 8a7888ec8b..8524e2aded (647072:647205)
Change log: 8a7888ec8b..8524e2aded
Full diff: 8a7888ec8b..8524e2aded

Changed dependencies
* src/base: 0b50d69249..4d7d3409f7
* src/build: 0797ede5ef..c2f2e5ab0f
* src/ios: e9d51ce300..b5be7e7906
* src/testing: a206cb6292..946f832584
* src/third_party: c51b0e9f82..8ae85677bd
* src/third_party/depot_tools: 422c432dd2..865445eb8a
* src/tools: 9247105d80..f05d4a5db9
DEPS diff: 8a7888ec8b..8524e2aded/DEPS

No update to Clang.

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

Change-Id: Iac84fa620a55ef9f8f352144783cc294d2055f78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131100
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@{#27438}
2019-04-03 10:36:29 +00:00
739506e45e Add thread safety annotations for some more PeerConnection members (part 12)
Plus all the annotations that were necessary to make things compile
again. I also had to send copies of some values owned by the signal
thread to the network thread, instead of letting the latter read them
itself.

Bug: webrtc:9987
Change-Id: Ic4b38696245584bab44956e60ac63753146e3ff4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131020
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27437}
2019-04-03 10:28:54 +00:00
c680c4a807 Revert "Running FrameBuffer on task queue."
This reverts commit 13943b7b7f6d00568912b9969db2c7871d18e21f.

Reason for revert: Breaks chromium import bots:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29

First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/2794

Original change's description:
> Running FrameBuffer on task queue.
> 
> This prepares for running WebRTC in simulated time where event::Wait
> based timing doesn't work.
> 
> Bug: webrtc:10365
> Change-Id: Ia0f9b1cc8e3c8c27a38e45b40487050a4699d8cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129962
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27422}

TBR=sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: I198a91ec1707cc8752a7fe55caf0f172e1b8e60a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131120
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27436}
2019-04-03 10:27:51 +00:00
fc6f3e5873 Include duration of pauses into sum of squared frames duration.
Bug: webrtc:10502
Change-Id: Ie905c0c9e8ca8fe07be585ce5a0d75e9eed6e865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130499
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27435}
2019-04-03 10:03:18 +00:00
78a5e96001 Reland "Add thread guards to JsepTransport"
This reverts commit caedb5db82b2bc8273910f4a0d1afb1d0e2994f3.

Reason for revert: Fixed issue (allow SetNeedsIceRestart from off-thread).

Original change's description:
> Revert "Add thread guards to JsepTransport"
>
> This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.
>
> Reason for revert: Breaks downstream.
>
> Original change's description:
> > Add thread guards to JsepTransport
> >
> > This ensures that JsepTransport's methods are either only accessed on the thread
> > that creates it, or using methods that are marked for off-thread use
> > (using a lock to prevent simultaneous access).
> >
> > The intent is to document the existing contract, and to make it easy to find the
> > actions needed to convert the class to a pure single-threaded class.
> >
> > Bug: webrtc:10300
> > Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27427}
>
> TBR=steveanton@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10300
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27429}

Change-Id: Ic32bfc04d96e657fc67c3d3999f77969e55ed994
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130962
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27434}
2019-04-03 09:41:38 +00:00
33d2a91737 Fix target bitrate RTCP messages behavior for SVC streams
This is a better solution than https://webrtc-review.googlesource.com/c/src/+/129929 (which got reverted).
This CL instead filters out unused SSRCs from RtpConfig for RtpVideoSender.

Bug: webrtc:10485
Change-Id: Iaa8d07681419a2387c8253eb38e08a0828e9e688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130505
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27433}
2019-04-03 09:36:38 +00:00
dbfb58b850 Ignore ERROR_ACCESS_DENIED when stopping Windows platform threads.
Wine implements ::QueueUserAPC incorrectly and returns
ERROR_ACCESS_DENIED when the thread is terminating instead of
ERROR_GEN_FAILURE. This is (hopefully) safe to do, assuming
the correct Windows implementation would never use ERROR_ACCESS_DENIED
in an actual failure case. I can't find documentation that says one
way or the other.

Bug: None
Change-Id: If74a40bb7e1cd49cc2266c31684bd88f1c667422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27432}
2019-04-03 09:08:52 +00:00
d9bf720c20 Add thread safety annotations for some more PeerConnection members (part 11)
After reviewer feedback, this CL was reduced to just adding scary
comments on two variables.

Bug: webrtc:9987
Change-Id: Id1e251ffd02e4ca8050235bd9f3971b5363f0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27431}
2019-04-03 09:05:41 +00:00
288cbe8200 Remove unused method in VCMInterFrameDelay.
Bug: none
Change-Id: I88f0f4011643736267f0dfa254cd65a936330253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130475
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27430}
2019-04-03 07:35:28 +00:00
caedb5db82 Revert "Add thread guards to JsepTransport"
This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.

Reason for revert: Breaks downstream.

Original change's description:
> Add thread guards to JsepTransport
> 
> This ensures that JsepTransport's methods are either only accessed on the thread
> that creates it, or using methods that are marked for off-thread use
> (using a lock to prevent simultaneous access).
> 
> The intent is to document the existing contract, and to make it easy to find the
> actions needed to convert the class to a pure single-threaded class.
> 
> Bug: webrtc:10300
> Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27427}

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

Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27429}
2019-04-03 07:15:38 +00:00