Commit Graph

25640 Commits

Author SHA1 Message Date
0ef117e14c Improving robustness of stable bandwidth estimate test.
It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.

Also fixing some minor test suite bug found during investigation.

Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
2019-01-25 15:06:17 +00:00
bebca61e5e Delete unused method SetSelectiveRetransmissions
Bug: None
Change-Id: I5a59b5776fe537ec380629f9e5e9ac98c9e1214b
Reviewed-on: https://webrtc-review.googlesource.com/c/119920
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26407}
2019-01-25 14:40:04 +00:00
728b5a4033 Fix initialization to prevent SIGSEGV
Bug: webrtc:10138
Change-Id: Ib299d2c5c08c07bbccf475b7e585cdd23830e238
Reviewed-on: https://webrtc-review.googlesource.com/c/119948
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26406}
2019-01-25 14:38:02 +00:00
b2d714110e Revert "Always use real VideoStreamsFactory in full stack tests"
This reverts commit 18cf2383aa2eb9de5778991c9d13b6b847143d37.

Reason for revert: Unexpected changes in webrtc_perf stats.

Original change's description:
> Always use real VideoStreamsFactory in full stack tests
> 
> Because quality scaling is enabled now in full stack test, correct
> factory should be used to compute actual resolution.
> 
> Also, since analyzed stream may be disabled completely now, change how
> analyzer considers the test finished --- count captured frames and
> stop if required amount of frames is captured and no new comparison were
> made.
> 
> Bug: webrtc:10204
> Change-Id: I205ebc892969ec1cf2d83e054e5c95e089d32104
> Reviewed-on: https://webrtc-review.googlesource.com/c/118687
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26358}

TBR=ilnik@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10204
Change-Id: Ia52fd55c9f68627166e0538d377003eae4ea518a
Reviewed-on: https://webrtc-review.googlesource.com/c/119946
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26405}
2019-01-25 14:27:10 +00:00
da37473a54 Make webrtc::ParseCandidate() public.
This is intended to be used in Blink to implement proper support
for the JavaScript RTCIceCandidate API.

Bug: chromium:683094
Change-Id: I93d117ef1bd9541593f2715bdf3291dc2941737f
Reviewed-on: https://webrtc-review.googlesource.com/c/119940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26404}
2019-01-25 13:58:57 +00:00
99ec6f39b9 AEC3: Remove unused kill-switches from AdjustConfig
Kill-switches removed:
WebRTC-Aec3UseShortDelayEstimatorWindow
WebRTC-Aec3ReverbBasedOnRenderKillSwitch
WebRTC-Aec3ReverbModellingKillSwitch
WebRTC-Aec3EnableUnityInitialRampupGain
WebRTC-Aec3EnableUnityNonZeroRampupGain
WebRTC-Aec3ShortReverbKillSwitch
WebRTC-Aec3NewFilterParamsKillSwitch
WebRTC-Aec3EnableLegacyDominantNearend
WebRTC-Aec3UseLegacyNormalSuppressorTuning
WebRTC-Aec3UseStationarityProperties
WebRTC-Aec3UseStationarityPropertiesAtInit
WebRTC-Aec3EarlyDelayDetectionKillSwitch

The change is tested for bit-exactness.

Bug: webrtc:8671
Change-Id: Ic7638002c0ca1bc5fc911e048285134c4df5d134
Reviewed-on: https://webrtc-review.googlesource.com/c/119921
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26403}
2019-01-25 13:37:13 +00:00
a9316c9445 frame_analyzer: exit with status 1 when video files fail to open
Bug: None
Change-Id: I6da6ee6d3686d97db63f09bd1cfa771ff1bdb403
Reviewed-on: https://webrtc-review.googlesource.com/c/119923
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26402}
2019-01-25 11:31:11 +00:00
a8f9e25778 Make sure lost packets are removed from FakeNetworkPipe.
Bug: webrtc:10239
Change-Id: I4391b35151c4cd99a2671a5126fd2546f82192ff
Reviewed-on: https://webrtc-review.googlesource.com/c/119641
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26401}
2019-01-25 08:57:45 +00:00
fe490d8e69 Roll chromium_revision b483b4fce1..6a5b2b19b1 (625914:626014)
Change log: b483b4fce1..6a5b2b19b1
Full diff: b483b4fce1..6a5b2b19b1

Changed dependencies
* src/base: 7998914884..9015adf2da
* src/ios: f7fa930347..528045cd2a
* src/testing: 0291324ebc..5ee5c49371
* src/third_party: dc9f88c901..bea3b73746
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/46e3f07075..1879ca54b9
* src/third_party/depot_tools: edfbc9ced2..80b9cf7dfd
* src/tools: ee9f8b35da..91febde900
DEPS diff: b483b4fce1..6a5b2b19b1/DEPS

No update to Clang.

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

Change-Id: Ia3658654cd96a674e11286add0e4449ba5a9c7de
Reviewed-on: https://webrtc-review.googlesource.com/c/119901
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@{#26400}
2019-01-25 08:33:12 +00:00
e47433f017 AEC3: Remove legacy render buffering
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.

Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
2019-01-25 08:31:12 +00:00
8a40edd802 Delete constant RTP_PAYLOAD_NAME_SIZE
Followup to cl https://webrtc-review.googlesource.com/c/src/+/119661

Bug: webrtc:6883
Change-Id: Ie3a06f7381a73b16fc5e7cd22366997cc95608ac
Reviewed-on: https://webrtc-review.googlesource.com/c/119760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26398}
2019-01-25 07:59:52 +00:00
76cf320110 Roll chromium_revision eedb2069ef..b483b4fce1 (625788:625914)
Change log: eedb2069ef..b483b4fce1
Full diff: eedb2069ef..b483b4fce1

Changed dependencies
* src/base: eba96e5cd1..7998914884
* src/build: 5ab04a69ba..018911f9a4
* src/ios: e0614546d7..f7fa930347
* src/testing: 137f694eb3..0291324ebc
* src/third_party: f7c2b7b838..dc9f88c901
* src/third_party/depot_tools: 4d965ee2d8..edfbc9ced2
* src/third_party/r8: D9fqCyfGhC3zMZFOE-4gzA0yox519Qd-DRgqnkqJuqgC..SlcbUnEufAQ-iuOwGOl8yYQuctmpf7bMqh59kBfpil0C
* src/tools: e02348e360..ee9f8b35da
DEPS diff: eedb2069ef..b483b4fce1/DEPS

No update to Clang.

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

Change-Id: I1e1fe9cb5c2a5ac72edfc11d77f0e47e5fa6d819
Reviewed-on: https://webrtc-review.googlesource.com/c/119841
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@{#26397}
2019-01-25 02:07:34 +00:00
b8c81c32ce Roll chromium_revision 3432970f4e..eedb2069ef (625619:625788)
Change log: 3432970f4e..eedb2069ef
Full diff: 3432970f4e..eedb2069ef

Changed dependencies
* src/base: 84bea49397..eba96e5cd1
* src/build: 59bf3c64e4..5ab04a69ba
* src/ios: 1826ede122..e0614546d7
* src/testing: 9074288d0b..137f694eb3
* src/third_party: 4e7fc335bc..f7c2b7b838
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b9dbf6c17c..46e3f07075
* src/third_party/depot_tools: 695e7cf352..4d965ee2d8
* src/tools: 66c76c7f23..e02348e360
DEPS diff: 3432970f4e..eedb2069ef/DEPS

No update to Clang.

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

Change-Id: I81e8ff630a03721dd835d781e5b16bc192db4245
Reviewed-on: https://webrtc-review.googlesource.com/c/119800
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@{#26396}
2019-01-24 21:03:22 +00:00
f50c6c2fb4 Introduce VideoQualityAnalyzerInjectionHelper.
VideoQualityAnalyzerInjectionHelper will be used to provide all required
entities to inject video quality analyzer into peer connection pipeline.

Bug: webrtc:10138
Change-Id: Iea7cf453311d809619839d5cf94b78a020ce9167
Reviewed-on: https://webrtc-review.googlesource.com/c/119642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26395}
2019-01-24 17:11:21 +00:00
3ea55d56eb Reland "Delete RtpUtility::Payload, and refactor RTPSender to not use it"
This is a reland of 171df9326200d1e01bce530e2ff01ac5890e6cb7

Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}

Tbr: danilchap@webrtc.org
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
2019-01-24 16:35:00 +00:00
5affbf2327 Turn off automatic quality scaling for simulcast in video_loopback.
The LibvpxVp8Encoder does not allow automatic quality scaling to be used when
encoding multiple resolutions (for simulcast).

Bug: None
Change-Id: Ic47d53850d03f399f80b6cf292fc607c19c1581d
Reviewed-on: https://webrtc-review.googlesource.com/c/119702
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26393}
2019-01-24 15:58:02 +00:00
3770b99e64 Allow repeated feedback packets in log parser.
Bug: webrtc:10170
Change-Id: I68cf729aa92b1266868f6ebcb211d9d4af031176
Reviewed-on: https://webrtc-review.googlesource.com/c/119300
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26392}
2019-01-24 14:29:15 +00:00
84ca69ad6e Add RTC event logging of LossNotification RTCP messages
Bug: webrtc:10226
Change-Id: Ib65970a8f13cd64529f3101993d40887168e313e
Reviewed-on: https://webrtc-review.googlesource.com/c/118933
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26391}
2019-01-24 13:33:57 +00:00
e2fffd7528 Roll chromium_revision 1aa6cb924c..3432970f4e (625210:625619)
Change log: 1aa6cb924c..3432970f4e
Full diff: 1aa6cb924c..3432970f4e

Changed dependencies
* src/base: b988df482e..84bea49397
* src/build: f2ca77c3aa..59bf3c64e4
* src/ios: 9a2b6d046d..1826ede122
* src/testing: b3ebc5e6e6..9074288d0b
* src/third_party: 629df88cb5..4e7fc335bc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/dd2de388fc..b9dbf6c17c
* src/third_party/depot_tools: f797143682..695e7cf352
* src/third_party/ffmpeg: 42bb040dde..4b75b8bab9
* src/third_party/harfbuzz-ng/src: 89bcfb204c..36fb2b4da9
* src/tools: a99484617c..66c76c7f23
DEPS diff: 1aa6cb924c..3432970f4e/DEPS

No update to Clang.

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

Change-Id: I5d718562a481561d0312092af4a8a6a6ad853e7d
Reviewed-on: https://webrtc-review.googlesource.com/c/119622
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@{#26390}
2019-01-24 13:32:37 +00:00
68d58602e2 Override default manifest from Chromium in WebRTC.
This allows rolling Chromium CL that removes API level 16 support:
https://chromium-review.googlesource.com/c/chromium/src/+/1423117

Bug: webrtc:10238, chromium:923477, chromium:922656
Change-Id: Icbed09256a4627dcae81230cd9a41a7f08c6a4d6
Reviewed-on: https://webrtc-review.googlesource.com/c/119580
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26389}
2019-01-24 13:00:49 +00:00
a67a9d9256 Handle zero number of spatial layers at calculation of VP9 SVC padding.
Bug: chromium:923330
Change-Id: I66e3b17e5a22b7de9d9b83d5dda486ec5b4364fc
Reviewed-on: https://webrtc-review.googlesource.com/c/119600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26388}
2019-01-24 12:38:12 +00:00
f8e7ccb967 Create new RTCP feedback message - LossIndication
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
   RTP sequence number. The decodability flag is true if and only if all of
   the frame's dependencies are known to be decodable, and the frame itself
   is not yet known to be unassemblable.
   * Clarification #1: In a multi-packet frame, the first packet's
     dependencies are known, but it is not yet known whether all parts
     of the current frame will be received.
   * Clarification #2: In a multi-packet frame, the dependencies would be
     unknown if the first packet was not received. Then, the packet will
     be known-unassemblable.

Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:21:00 +00:00
2d0505017a Revert "Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)"
This reverts commit 88ca008e56eaf3c0986e878b60cf986c77b993f2.

Reason for revert: Fails with Android lint errors in post-submit

Original change's description:
> Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)
> 
> Change log: 1aa6cb924c..faaba5b0a8
> Full diff: 1aa6cb924c..faaba5b0a8
> 
> Changed dependencies
> * src/base: b988df482e..84bea49397
> * src/build: f2ca77c3aa..59bf3c64e4
> * src/ios: 9a2b6d046d..90e80ed872
> * src/testing: b3ebc5e6e6..9074288d0b
> * src/third_party: 629df88cb5..fdaf600d3b
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/dd2de388fc..b9dbf6c17c
> * src/third_party/depot_tools: f797143682..695e7cf352
> * src/third_party/ffmpeg: 42bb040dde..4b75b8bab9
> * src/third_party/harfbuzz-ng/src: 89bcfb204c..36fb2b4da9
> * src/tools: a99484617c..ddf43e8bc7
> DEPS diff: 1aa6cb924c..faaba5b0a8/DEPS
> 
> No update to Clang.
> 
> TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
> BUG=None
> 
> Change-Id: Id449c0b73caf61c2c5fa30e6c5f85795483602af
> Reviewed-on: https://webrtc-review.googlesource.com/c/119620
> 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@{#26382}

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com

Change-Id: Iaab986749602ec489ef7b3748ec658484c12b67d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/119660
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26386}
2019-01-24 12:06:17 +00:00
81d4bf7af6 Revert "Delete RtpUtility::Payload, and refactor RTPSender to not use it"
This reverts commit 171df9326200d1e01bce530e2ff01ac5890e6cb7.

Reason for revert: Breaks downstream project

Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
> 
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
> 
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}

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

Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
2019-01-24 12:02:12 +00:00
1e27fec293 Negate flag name for prerender smoothing and update comments.
Further, strictly require VideoReceiveStream::Config::rendererer
to be non-null when the VideoReceiveStream is started. This is
already true by construction in the production code.

Bug: None
Change-Id: Ia0a41cfafa44215efc195a9eb6204194930c3dde
Reviewed-on: https://webrtc-review.googlesource.com/c/115040
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26384}
2019-01-24 11:53:26 +00:00
2fd09a40af Remove deprecated code from audio device.
Bug: webrtc:7306, webrtc:10198
Change-Id: Iaeef4d7449c18325511f1763eba510b385959bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/118446
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26383}
2019-01-24 11:27:38 +00:00
88ca008e56 Roll chromium_revision 1aa6cb924c..faaba5b0a8 (625210:625596)
Change log: 1aa6cb924c..faaba5b0a8
Full diff: 1aa6cb924c..faaba5b0a8

Changed dependencies
* src/base: b988df482e..84bea49397
* src/build: f2ca77c3aa..59bf3c64e4
* src/ios: 9a2b6d046d..90e80ed872
* src/testing: b3ebc5e6e6..9074288d0b
* src/third_party: 629df88cb5..fdaf600d3b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/dd2de388fc..b9dbf6c17c
* src/third_party/depot_tools: f797143682..695e7cf352
* src/third_party/ffmpeg: 42bb040dde..4b75b8bab9
* src/third_party/harfbuzz-ng/src: 89bcfb204c..36fb2b4da9
* src/tools: a99484617c..ddf43e8bc7
DEPS diff: 1aa6cb924c..faaba5b0a8/DEPS

No update to Clang.

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

Change-Id: Id449c0b73caf61c2c5fa30e6c5f85795483602af
Reviewed-on: https://webrtc-review.googlesource.com/c/119620
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@{#26382}
2019-01-24 11:26:32 +00:00
fc2175da73 Introduce QualityAnalyzingVideoEncoder and QualityAnalyzingVideoDecoder.
This encoder will be used to inject VideoQualityAnalyzerInterface into
VideoEncoder, so it will be able to measure its metrics and also trace
frames from capturing on one peer side to rendering on another peer side.
The decoder will be used for the same purpose but in VideoDecoder pert.

Bug: webrtc:10138
Change-Id: Idf719753e3c0b3b1369ff206365bf0558705eb98
Reviewed-on: https://webrtc-review.googlesource.com/c/117363
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26381}
2019-01-24 11:15:12 +00:00
171df93262 Delete RtpUtility::Payload, and refactor RTPSender to not use it
Replaced by a payload type --> video codec map in RTPSenderVideo,
where it is used to select the right packetizer.

Bug: webrtc:6883
Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119263
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26380}
2019-01-24 10:47:21 +00:00
2820d174b5 Roll chromium_revision 1cac36a781..1aa6cb924c (624101:625210)
Change log: 1cac36a781..1aa6cb924c
Full diff: 1cac36a781..1aa6cb924c

Changed dependencies
* src/base: 66f195cc35..b988df482e
* src/build: 13c868f507..f2ca77c3aa
* src/buildtools: 40194ab039..2f02e1f363
* src/ios: 296c1e46f1..9a2b6d046d
* src/testing: cc93e4d879..b3ebc5e6e6
* src/third_party: 242cd92ba3..629df88cb5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8e77731b13..dd2de388fc
* src/third_party/depot_tools: deab113bfb..f797143682
* src/third_party/googletest/src: 879ac092fd..9518a57428
* src/third_party/harfbuzz-ng/src: 0d2727f4fe..89bcfb204c
* src/tools: 888c147a20..a99484617c
DEPS diff: 1cac36a781..1aa6cb924c/DEPS

Clang version changed 350768:351477
Details: 1cac36a781..1aa6cb924c/tools/clang/scripts/update.py

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

Change-Id: Iddbb27c20de141c630adacdfafc1d76ad9aade10
Reviewed-on: https://webrtc-review.googlesource.com/c/119283
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26379}
2019-01-24 09:15:59 +00:00
dbb49dfb27 Moving UniqueIdGenerator into rtc_base.
UniqueIdGenerator classes are useful outside the pc directory.
This change moves them to the rtc_base directory to enable code
in all directories to reference them.

Bug: None
Change-Id: I1c77da87ea26d9611f37dc1d4d2c16006a6589c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26378}
2019-01-24 00:52:31 +00:00
6fde78cb2f Prevent mac_framework_bundle configs from getting reset
If the invoker sets "configs", the default configs would be overridden
by the forward_variables_from() call.  This fixes linker errors for
certain shared libraries on Mac Asan when rolling in the change that
switches Mac to build with our in-tree libc++.

BUG=chromium:924121
R=yvesg
CC=thakis

Change-Id: If9db41b724f891034086c64dc7ba38a6406aef92
Reviewed-on: https://webrtc-review.googlesource.com/c/119286
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Tom Anderson <thomasanderson@google.com>
Cr-Commit-Position: refs/heads/master@{#26377}
2019-01-23 19:21:46 +00:00
ce7032b884 Fixing snake_case files that were renamed in PRESUBMIT.py
In the great rename of 2019, several files were renamed to snake_case.
Some of these files were referenced in PRESUBMIT.py alas, the references
were not updated.

Bug: webrtc:10159
Change-Id: I462938a1e8ea11b865d9427b40cba43fd0aab998
Reviewed-on: https://webrtc-review.googlesource.com/c/119160
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26376}
2019-01-23 18:38:53 +00:00
6a32de4285 Fix potential race in CallTest.
The FrameGeneratorCapturer instances continue to live after
RunBaseTest() returns, and have their own internal task queues. This
means any class that listens for frames may be called after return
from RunBaseTest(), at which point they may be destroyed.

This CL makes sure we remove any capturer before returning.

A specific example of this problem is
VideoSendStreamTest.SuspendBelowMinBitrate

Bug: None
Change-Id: I857566301acce3e32c0888c7a1d2ee3470e6eb28
Reviewed-on: https://webrtc-review.googlesource.com/c/116684
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26375}
2019-01-23 16:49:55 +00:00
2c58ba1f24 Move simulcast hysteresis factor parsing to RateControlSettings
Bug: webrtc:10223
Change-Id: I962ca959afbcd8c27a0f79533c6e3c97369c697e
Reviewed-on: https://webrtc-review.googlesource.com/c/119262
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26374}
2019-01-23 16:34:34 +00:00
83d5e86163 Use EncoderSimulcastProxy for all codecs
Some codecs don't support directly creating simulcast layers with
non-optimal parameters. This proxy will detect this and create
multiple encoders then, one for each layer as a fallback.

Bug: webrtc:10069
Change-Id: I4bcafcfdd68d9ed466e2fafe564db849de6ed4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/119264
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26373}
2019-01-23 15:59:54 +00:00
b599787969 Make UlpfecReceiverImpl use rtc::TimeMillis, not Clock::GetRealTimeClock
Bug: webrtc:6733
Change-Id: I0cdfc781ff0daff18d1fc0b6243fb1f95f704cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/119220
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26372}
2019-01-23 14:54:08 +00:00
4b4266f00f Move parsing of trusted rate controller to RateControlSettings
Bug: webrtc:10223
Change-Id: Iadf46e278e0f994ed95ff1a240c1f39a0421ab7c
Reviewed-on: https://webrtc-review.googlesource.com/c/119261
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26371}
2019-01-23 14:37:08 +00:00
470a5eae93 Introduces common AudioAllocationSettings class.
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.

Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
2019-01-23 12:13:29 +00:00
33b716f7dd Publish task queue test suite.
The set of tests is a copy of rtc_base/task_queue_unittests excluding tests
that verify rtc::NewClosure rather than task queue implementation itself.

Bug: webrtc:10191
Change-Id: I94e962ad63ff6510c43a97ef0cd4da7d08f25538
Reviewed-on: https://webrtc-review.googlesource.com/c/118445
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26369}
2019-01-23 11:55:12 +00:00
b0397d69a9 Always send abs-send-time when negotiated and do not filter it out.
Previously, when abs-send-time was negotiated, it was not sent if TWCC
was enabled. With this FieldTrial, abs-send-time header extension is
sent even if TWCC was negotiated in addition to abs-send-time.

Bug: webrtc:10234
Change-Id: I3af85720760882e89760888d43996fe85def619a
Reviewed-on: https://webrtc-review.googlesource.com/c/118936
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26368}
2019-01-23 11:30:39 +00:00
ae6e0b2058 [CodeHealth] Fix use after std::move instances.
The parameter was expanded twice in the macro,
leading to double use and move.
This is both an example of:
 * Issue spotted by clandtidy's bug-prone patterns.
 * Premature optimization.

Bug: webrtc:9855
Change-Id: I1a0cb2c99f95c6aec79ba1eb198aa39743ccbcd9
Reviewed-on: https://webrtc-review.googlesource.com/c/119042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26367}
2019-01-23 11:04:11 +00:00
e54287a4ae Correctly specify Mac version as 10.13 for iOS simulator tests
According to breaking change in https://chromium-review.googlesource.com/c/chromium/tools/build/+/1422562
Similarly to other configs: https://cs.chromium.org/search/?q=Mac-10.1+f:ios+f:json

Bug: webrtc:10236
Change-Id: Ia03500a057de784336b225225053dc37dcb6e761
Reviewed-on: https://webrtc-review.googlesource.com/c/119222
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26366}
2019-01-23 10:32:57 +00:00
df919fb425 Don't pretend we've received an end-of-candidates indication.
Since end-of-candidates signalling isn't implemented yet, the ice transport shouldn't reach completed. We also shouldn't assume that the transport has failed because gathering is complete without candidates, as we might still get remote candidates.

Bug: chromium:922588
Change-Id: I332f57be494efc775819d80908e9f39610311f82
Reviewed-on: https://webrtc-review.googlesource.com/c/118741
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26365}
2019-01-23 10:16:43 +00:00
28522dc6e3 Rename new build targets to follow the recent large file rename
Bug: webrtc:9987, webrtc:10159
Change-Id: I7f56913e81bce0b5e1f05b8c3e8b848870f12f44
Reviewed-on: https://webrtc-review.googlesource.com/c/118937
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26364}
2019-01-23 10:04:43 +00:00
79f0d4d0c7 Enables feature to account for unacknowledged data.
By enabling this trial, we can also remove reporting of packet
feedback status from send streams that was used before.

Bug: webrtc:9718
Change-Id: I3e7c4656b0ac6592a834617e044f23a072454181
Reviewed-on: https://webrtc-review.googlesource.com/c/118281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26363}
2019-01-23 10:00:52 +00:00
7e4341db60 Reland "Reland "Adds richer packet and ice processing to ParsedRtcEventLog.""
This is a reland of 6fc6a0cbb10ee0e988b47f48935b630ba41d109d

Original change's description:
> Reland "Adds richer packet and ice processing to ParsedRtcEventLog."
> 
> This is a reland of 4306a25dfcaba7defe09f5d4b669736d374fe985
> 
> Original change's description:
> > Adds richer packet and ice processing to ParsedRtcEventLog.
> > 
> > Bug: webrtc:10170
> > Change-Id: I0f10a8c0b5656917a806cf0f3ad88b7a6baee000
> > Reviewed-on: https://webrtc-review.googlesource.com/c/116069
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26268}
> 
> Bug: webrtc:10170
> Change-Id: Ie523427acba02b554583223b9ef800249d8d8f2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/117724
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26350}

Bug: webrtc:10170
Change-Id: I9b57ca754197822de9966ee4c93526c7f2159dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/118784
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26362}
2019-01-23 10:00:01 +00:00
41dd0bc4e5 Fix typo in rtc_base/thread_checker.h.
Bug: None
No-Try: True
Change-Id: Ie5562b3af54c2ce95d7433ba4237f976a2e60df3
Reviewed-on: https://webrtc-review.googlesource.com/c/119040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26361}
2019-01-22 23:32:44 +00:00
067dc86c8a Make SetFirstSubFrameInFrame and SetLastSubFrameInFrame protected
These methods should only be used when parsing frames produced
by an older client; newer clients should not attempt to set
these values.

(When talking to older clients, TRUE is hard-coded. When talking
to newer clients, these flags are deprecated.)

Bug: webrtc:10214
Change-Id: I8537869ef3112f4ce9531c6becc33951715685a1
Reviewed-on: https://webrtc-review.googlesource.com/c/118421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26360}
2019-01-22 12:32:47 +00:00
3fdf90d621 PSFB without REMB magic word is not an error
Several PSFB messages might be supported, distinguished using
the unique identifier. If the unique identifier is not REMB, it's
not an error, and so a warning should not be issued.

Bug: webrtc:10226
Change-Id: I5e79b473bd54cf0964f19329efb33354f63f5d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/118686
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26359}
2019-01-22 12:29:47 +00:00