Commit Graph

25551 Commits

Author SHA1 Message Date
18f0c3c038 Add RegisterAudioSendPayload() method
In preparation of removing CodecInst.

Bug: webrtc:7626
Change-Id: I8955d17dbb3ec15177e505ae420376b542d48410
Reviewed-on: https://webrtc-review.googlesource.com/c/113306
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25919}
2018-12-06 12:44:53 +00:00
d8a1b7a5c5 Use opaque int as payload_type in MediaTransportInterface
Replaces enum VideoCodecType for video frames and uint8_t for audio
frames.

Also delete method
MediaTransportVideoSinkInterface::OnKeyFrameRequested; it needs to be
added as a send-side interface instead (for a later cl).

Bug: webrtc:9719
Change-Id: I2cfdbacc267afc75c448512e2cc6de0ec9966a2d
Reviewed-on: https://webrtc-review.googlesource.com/c/113180
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25918}
2018-12-06 12:37:27 +00:00
5658ea660e Fix iOS version for internal.client.webrtc perf test
This adjusts iOS version to the actual one on the tester bot.

Bug: webrtc:10047
Change-Id: I7d104f331450192142c8c2c1259a3207dcee45ed
Reviewed-on: https://webrtc-review.googlesource.com/c/113420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25917}
2018-12-06 12:26:47 +00:00
a728c919a4 Fix performance bug in CreateSendSideBweSimulationGraph
Bug: webrtc:10097
Change-Id: Ie60619084cd4bd47f5f81d06262ba62631eac12f
Reviewed-on: https://webrtc-review.googlesource.com/c/113423
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25916}
2018-12-06 12:16:36 +00:00
628f37a6fe Delete a cricket::DtlsTransport when PC is closed
This avoids use-after-free problems that occur when references
to webrtc::DtlsTransport objects are held outside of the PC.

Bug: chromium:907849
Change-Id: Id428c8e616482eff0f4327d2eac17e29bb3f6484
Reviewed-on: https://webrtc-review.googlesource.com/c/113303
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25915}
2018-12-06 12:06:34 +00:00
fafae11bfc Allow parsing RTC event logs without a transaction ID.
Bug: webrtc:9972
Change-Id: I01d60671d249adbd55f25c8f49f205b18787cbf4
Reviewed-on: https://webrtc-review.googlesource.com/c/113304
Reviewed-by: Zach Stein <zstein@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25914}
2018-12-06 09:59:45 +00:00
518add35a8 Roll chromium_revision 68c6b982a7..d6dec3971c (614157:614285)
Change log: 68c6b982a7..d6dec3971c
Full diff: 68c6b982a7..d6dec3971c

Changed dependencies
* src/base: 8778f36ffb..1bc039647f
* src/build: f8704051e5..84f0bf98ad
* src/ios: f2a06afe6e..11779ae7d1
* src/testing: 92b188b3ea..897a09fa69
* src/third_party: 03c7212525..6d9122ca49
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fa0beb5bf0..c7cc237f95
* src/third_party/depot_tools: 19238fc343..2e00228777
* src/tools: 4589892399..1c79b0fc32
DEPS diff: 68c6b982a7..d6dec3971c/DEPS

No update to Clang.

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

Change-Id: Idac08c10b9de476dd5243e8de6e5cf8d883e5d1e
Reviewed-on: https://webrtc-review.googlesource.com/c/113383
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@{#25913}
2018-12-06 08:22:11 +00:00
5f93a3eaa9 Roll chromium_revision 48afe1fce1..68c6b982a7 (614046:614157)
Change log: 48afe1fce1..68c6b982a7
Full diff: 48afe1fce1..68c6b982a7

Changed dependencies
* src/base: 38c52177ea..8778f36ffb
* src/ios: ffc1609b07..f2a06afe6e
* src/testing: 3f5136437f..92b188b3ea
* src/third_party: af23677231..03c7212525
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/58f298110d..fa0beb5bf0
* src/tools: 10a524fa51..4589892399
DEPS diff: 48afe1fce1..68c6b982a7/DEPS

No update to Clang.

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

Change-Id: Iefb2e2dd3376c7e13726e42003d234d02fbae214
Reviewed-on: https://webrtc-review.googlesource.com/c/113326
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@{#25912}
2018-12-05 23:37:41 +00:00
c60a77731d Process RTP before RTCP in RTC event log analyzer.
This handles an unlikely corner case where you receive a RTCP feedback for a packet the same millisecond that you send it.

Bug: None
Change-Id: I77f460bef4073d4d9c5633c88f4d2dd8470f8577
Reviewed-on: https://webrtc-review.googlesource.com/c/113305
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25911}
2018-12-05 21:04:55 +00:00
75726f2852 Roll chromium_revision 92bac9a669..48afe1fce1 (613922:614046)
Change log: 92bac9a669..48afe1fce1
Full diff: 92bac9a669..48afe1fce1

Changed dependencies
* src/base: 23fe4e6235..38c52177ea
* src/ios: b945749f3b..ffc1609b07
* src/testing: a5684e641c..3f5136437f
* src/third_party: cc7029ba2b..af23677231
* src/third_party/depot_tools: 687ca907fd..19238fc343
* src/third_party/harfbuzz-ng/src: 000d4b128e..79e7e3445e
* src/third_party/r8: ndmKWh0vZhDc2iLXEETOuWXVfafHbqwI_FcSgJJIfpoC..gMAAlElX8RMw__5KOpk-Ckdx3XDyEXspJVslmnblsrgC
* src/tools: cf5c7c5fbd..10a524fa51
DEPS diff: 92bac9a669..48afe1fce1/DEPS

No update to Clang.

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

Change-Id: Ibce718cf581f1626398c0db6620e8e15f8cd7041
Reviewed-on: https://webrtc-review.googlesource.com/c/113322
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@{#25910}
2018-12-05 19:41:15 +00:00
4a817f6e6f Store timestamp of first and last event in RTC event log parser.
This was previously only done for the legacy wire format.

Bug: webrtc:8111
Change-Id: I82767c8eafb35a50967c6ff3d0131c3981957c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/112590
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25909}
2018-12-05 17:05:43 +00:00
bd2cf71865 Remove functions to inject a TaskQueue in RtcEventLog creation.
The event log implementation will be simpler if it creates its own TaskQueue.
If we really need the "injectable" functionality, it could be achieved via a
TaskQueueFactory that returns a move-constructible TaskQueue.

Bug: webrtc:10085
Change-Id: I538be3dd77c09be2f5bae015227067acd6af8355
Reviewed-on: https://webrtc-review.googlesource.com/c/113140
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25908}
2018-12-05 16:35:04 +00:00
b438b5a33d Reland "Change ReceiveStatistics reaction to large sequence numbers jumps"
This reverts commit 7e0299e2452b021fcd14a8fdb86257459eeacf90.

Reason for revert: audio receive stream fix not to use 0 reordering threshold

Original change's description:
> Revert "Change ReceiveStatistics reaction to large sequence numbers jumps"
> 
> This reverts commit c4f120130f495e9726bf221356642de69125f4a2.
> 
> Reason for revert: breaks downstream tests due to zero max reordering for audio receive channels
> 
> Original change's description:
> > Change ReceiveStatistics reaction to large sequence numbers jumps
> > 
> > Consider stream restart when two sequential packets arrived far from
> > previous packets' sequence numbers.
> > instead of resetting on single one.
> > For packet loss calculation ignore sequence number gap during reset.
> > 
> > Bug: webrtc:9445, b/38179459
> > Change-Id: I0c2717ef8f9ec182b280ae757b5582f56d9afcef
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111962
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25890}
> 
> TBR=danilchap@webrtc.org,asapersson@webrtc.org
> 
> Change-Id: Icc9f4d86d9f0b07f0fa2f3d443f9a90aa91f5e21
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9445, b/38179459
> Reviewed-on: https://webrtc-review.googlesource.com/c/113067
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25897}

TBR=danilchap@webrtc.org,asapersson@webrtc.org

Change-Id: I8747aa5cb6209b92fafefed077bc19d305d11db6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9445, b/38179459
Reviewed-on: https://webrtc-review.googlesource.com/c/113263
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25907}
2018-12-05 16:31:00 +00:00
dc107965bd Fix AGC2 fuzzer coverage.
Bug: webrtc:10084
Change-Id: Icc51994fe5ab16188c41452e887cbe7a6b8b9aff
Reviewed-on: https://webrtc-review.googlesource.com/c/112941
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25906}
2018-12-05 15:55:42 +00:00
2a977cf466 For audio receive channel use default max reordering threshold instead of 0
setting max reordering recently has been fix to actually set it.
(https://webrtc-review.googlesource.com/c/src/+/111752)
Another recent change fix stats to skip counting large sequence number jumps as packet loss
(https://webrtc-review.googlesource.com/c/src/+/111962)

max reordering thresholds affects how packet loss is calculated.
Packet loss is then reported to remote sending participant in rtcp receiver reports.
Sender uses packet loss mostly for stats, but also e.g. for opus fec adjustment.

Setting threshold to zero de-facto imply all packets should be considered in order.
That bug was mitigated by two other bugs mentioned above

This change increase threshold to default 50 packets aligning it with Video receiver
and unblocks (re)landing 2nd fix

Bug: b/120482366
Change-Id: Iadda0c2148ed84dd83c01183cfe9285568db4e29
Reviewed-on: https://webrtc-review.googlesource.com/c/113064
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25905}
2018-12-05 14:52:23 +00:00
c979c46bda Re-add MSVC debug bots to CQ
This reverts commit e80e0132c165b604e7a6684b356cddd857a60f7e.

Compilation has been fixed in https://chromium-review.googlesource.com/c/chromium/src/+/1297536

No-Try: True
Bug: webrtc:9695, webrtc:10071
Change-Id: I0e5b657c9197f1b71c2473aae366f06b0daad3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/113220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25904}
2018-12-05 12:20:56 +00:00
b819ed6df0 Roll chromium_revision 2e285ebae2..92bac9a669 (613019:613922)
Change log: 2e285ebae2..92bac9a669
Full diff: 2e285ebae2..92bac9a669

Changed dependencies
* src/base: 62febbdbd7..23fe4e6235
* src/build: 8b1ff06550..f8704051e5
* src/ios: 2c8e8f83db..b945749f3b
* src/testing: da3cc6c84a..a5684e641c
* src/third_party: a862efe9b4..cc7029ba2b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6f862e54f2..58f298110d
* src/third_party/depot_tools: 0b287c5bca..687ca907fd
* src/third_party/harfbuzz-ng/src: e0307de818..000d4b128e
* src/tools: cc443eb2fd..cf5c7c5fbd
DEPS diff: 2e285ebae2..92bac9a669/DEPS

No update to Clang.

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

Change-Id: Ia5d418b0ca23b47f4ef34b163cf32a429e061711
Reviewed-on: https://webrtc-review.googlesource.com/c/113162
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25903}
2018-12-05 10:44:17 +00:00
657b296ff5 Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
2018-12-05 10:38:23 +00:00
5b1477839d [Unified Plan] If "a=msid" is missing, create default stream.
Prior to this CL, if the "a=msid" attribute was missing it was treated
the same as if "no streams" were explicitly signaled (a=msid:-); the
receivers would not be associated with any streams.

In order to support legacy endpoints that don't recognize "a=msid" that
assume the Plan B behavior of a stream being created anyway, this CL
creates a stream with a random ID in such cases. For background, see
https://github.com/web-platform-tests/wpt/pull/14054.

Bug: chromium:907508
Change-Id: I9d9dd0e4ba8f9941f8652f4d7873adc560777cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/112900
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25901}
2018-12-05 09:53:21 +00:00
e5e36ddc40 Roll chromium_revision 3546854f59..2e285ebae2 (612694:613019) + fix JNI
This changelist is based on Chromium autoroller CL
https://webrtc-review.googlesource.com/c/src/+/112847
with additional JNI fixes needed to propagate upstream changes
introduced in
c99e905516


Change log: 3546854f59..2e285ebae2
Full diff: 3546854f59..2e285ebae2

Changed dependencies
* src/base: 0551460b2b..62febbdbd7
* src/build: 59f4bb0792..8b1ff06550
* src/ios: 0c78d113b3..2c8e8f83db
* src/testing: d387a4a97a..da3cc6c84a
* src/third_party: e31ab38349..a862efe9b4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1b98245e3c..6f862e54f2
* src/third_party/depot_tools: 016601cc21..0b287c5bca
* src/third_party/r8: uM1IGlYVeBYwmhwRCSMVqRvmu4YFlL7M2yLwZ1DWUvAC..ndmKWh0vZhDc2iLXEETOuWXVfafHbqwI_FcSgJJIfpoC
* src/tools: 476768d37c..cc443eb2fd
DEPS diff: 3546854f59..2e285ebae2/DEPS

No update to Clang.

No-Try: True
Bug: chromium:898660
Change-Id: I8be89e16d9639d96fc09f053e29414381a486846
Reviewed-on: https://webrtc-review.googlesource.com/c/112595
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25900}
2018-12-05 09:48:51 +00:00
04744aee2f Set priority of iOS test tasks to 30
The default Swarming priority is 200 but it's recommended to raise it.
Chrome's tasks are set to 30, and that can cause our tasks to be discarded.

Bug: chromium:911787
Change-Id: Ied5eed4bc37890ede6c29d2fd743e102f5622d11
Reviewed-on: https://webrtc-review.googlesource.com/c/113145
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25899}
2018-12-05 08:59:44 +00:00
a713505c18 Preserve timestamp in DTLS writable event.
Bug: webrtc:10088
Change-Id: I63cd8cd3142c466c4ade74caee231a50fee8191e
Reviewed-on: https://webrtc-review.googlesource.com/c/113066
Reviewed-by: Zach Stein <zstein@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25898}
2018-12-04 20:49:11 +00:00
7e0299e245 Revert "Change ReceiveStatistics reaction to large sequence numbers jumps"
This reverts commit c4f120130f495e9726bf221356642de69125f4a2.

Reason for revert: breaks downstream tests due to zero max reordering for audio receive channels

Original change's description:
> Change ReceiveStatistics reaction to large sequence numbers jumps
> 
> Consider stream restart when two sequential packets arrived far from
> previous packets' sequence numbers.
> instead of resetting on single one.
> For packet loss calculation ignore sequence number gap during reset.
> 
> Bug: webrtc:9445, b/38179459
> Change-Id: I0c2717ef8f9ec182b280ae757b5582f56d9afcef
> Reviewed-on: https://webrtc-review.googlesource.com/c/111962
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25890}

TBR=danilchap@webrtc.org,asapersson@webrtc.org

Change-Id: Icc9f4d86d9f0b07f0fa2f3d443f9a90aa91f5e21
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9445, b/38179459
Reviewed-on: https://webrtc-review.googlesource.com/c/113067
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25897}
2018-12-04 17:16:22 +00:00
6353fbc237 Offset IceCandidatePairEventType enum in event log visualizer
Bug: NONE
Change-Id: I17b586702d02a49917e707788caf091955c9357c
Reviewed-on: https://webrtc-review.googlesource.com/c/112723
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25896}
2018-12-04 16:50:43 +00:00
5546aef682 Vp9 flexible mode fixes
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
  and return several frames combined from FrameBuffer.

Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
2018-12-04 15:36:28 +00:00
77894ccb5d New method EncodedImage::set_buffer
This setter method is intended to replace all direct assignments to
the |_size| member. A later cl will make that member private, and
rename it to |capacity_|.

Bug: webrtc:9378
Change-Id: I37e9eb54d1c72bcd4cb8a1cfef34bbc6c209bd0d
Reviewed-on: https://webrtc-review.googlesource.com/c/113060
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25894}
2018-12-04 15:25:55 +00:00
ebad1770ab Include event_wrapper.h only where used.
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.

Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
2018-12-04 14:50:18 +00:00
f5997c9bae Delete unused member RTPSender::last_capture_time_ms_sent_
It was updated, but otherwise unused. And in addition, the update code
lacked needed synchronization.

Bug: webrtc:10033
Change-Id: I2a7b45550543a75d5f6b53032b512fd2fd120290
Reviewed-on: https://webrtc-review.googlesource.com/c/113041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25892}
2018-12-04 13:35:03 +00:00
af2adda252 Explicit comparisons on NetworkRoute.
Since not all fields are compared on NetworkRoute structs, the ==
operator overload doesn't really make the code easier to read. In fact
the feature that it only compares a subset of the fields is only used
once, at the other places, all fields are compared.

Removing the overload makes it more clear what is compared at each call
site.

Bug: webrtc:9883
Change-Id: I74f7eb32b602aa33fd282a815b71a172ae3f6a8b
Reviewed-on: https://webrtc-review.googlesource.com/c/113001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25891}
2018-12-04 12:36:50 +00:00
c4f120130f Change ReceiveStatistics reaction to large sequence numbers jumps
Consider stream restart when two sequential packets arrived far from
previous packets' sequence numbers.
instead of resetting on single one.
For packet loss calculation ignore sequence number gap during reset.

Bug: webrtc:9445, b/38179459
Change-Id: I0c2717ef8f9ec182b280ae757b5582f56d9afcef
Reviewed-on: https://webrtc-review.googlesource.com/c/111962
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25890}
2018-12-04 12:16:49 +00:00
ee59a17bbb Ensure GenericFrameDescriptor byte representation match
on receiver and on sender

Bug: webrtc:9795
Change-Id: I13dfcc6dce0be55d90406e27ba855ef5e786af24
Reviewed-on: https://webrtc-review.googlesource.com/c/104900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25889}
2018-12-04 11:52:25 +00:00
74fb822b67 Allow probing up to 2x allocation limit
The limit we put on probing is a bit too conservative now. If an
allocation limit is set, this CL allows probing up to 2x the current
max allocation limit.

This better handles overshooting when networks actually have the
capacity to allow bursts.

Bug: webrtc:10070
Change-Id: I0003f6b22512c13b6a83c1934952a2c3a2b70b48
Reviewed-on: https://webrtc-review.googlesource.com/c/112905
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25888}
2018-12-04 11:43:54 +00:00
9cccf85314 Delete unused video coding error codes.
Bug: webrtc:8064
Change-Id: I0fb714acc1bd86e230139eb929137a10710652ba
Reviewed-on: https://webrtc-review.googlesource.com/c/113000
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25887}
2018-12-04 11:39:34 +00:00
ebb677773a Allow access to specific RTCP types in new format event log parser.
Bug: webrtc:8111
Change-Id: Ia727c8fd9ddaa9f0c17fea5af8f544ad01e66367
Reviewed-on: https://webrtc-review.googlesource.com/c/112382
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25886}
2018-12-04 10:49:21 +00:00
41390474fc Add LookupDtlsTransportByMid to proxy map
Bug: chromium:907849
Change-Id: Ib707f18fdf67617967b83cd17f0454d182b8011f
Reviewed-on: https://webrtc-review.googlesource.com/c/112907
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25885}
2018-12-04 10:29:32 +00:00
d0b69a8c50 Send and receive color space information if available
Bug: webrtc:8651
Change-Id: I244647cb1ccbda66fce83ae925cf4273c5a6568b
Reviewed-on: https://webrtc-review.googlesource.com/c/112383
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25884}
2018-12-03 21:07:45 +00:00
a201204215 Adding SDP parsing for Simulcast.
Parsing simulcast according to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-simulcast-13#section-5.1
Created SdpSerializer for making serialized components more testable.
Simulcast functionality is still not accessible to users.

Bug: webrtc:10055
Change-Id: Ia6e4cef756cb954521dd19e22911f8eb6498880e
Reviewed-on: https://webrtc-review.googlesource.com/c/112160
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25883}
2018-12-03 20:13:53 +00:00
ec086d842c Fixes race in SendSideCongestionControllerTest.OldFeedback.
This fixes an issue where SendSideCongestionControllerTest.OldFeedback
calls a function that posts a task on a TaskQueue and immediately after
changes the mocked observer that is called from that task.

Bug: webrtc:10056
Change-Id: Ib1cca5bf695482e75106bfc715662e4f76c381d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112940
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25882}
2018-12-03 17:26:32 +00:00
ec0f45be11 Revert "Remove CodecInst pt.1"
This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851.

Reason for revert: breaks downstream

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

TBR=solenberg@webrtc.org,kwiberg@webrtc.org

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}
2018-12-03 15:50:51 +00:00
9d54bd8898 AEC3: Fix ENR threshold for WebRTC-Aec3UseLegacyNormalSuppressorTuning
Fixes the ENR threshold used in the dominant nearend detection when
the kill-switch WebRTC-Aec3UseLegacyNormalSuppressorTuning is pulled.

Bug: webrtc:8671,chromium:911141
Change-Id: I30ee58009633b3a9e12eff692226baada624a049
Reviewed-on: https://webrtc-review.googlesource.com/c/112903
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25880}
2018-12-03 15:19:00 +00:00
056f9738bf Remove CodecInst pt.1
Update audio_coding tests to not use CodecInst.

Bug: webrtc:7626
Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
Reviewed-on: https://webrtc-review.googlesource.com/c/112594
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25879}
2018-12-03 15:16:20 +00:00
450b548cad Encode frames at the largest layer resolution instead of last layer
The current code assumes that layers are ordered from the smallest
to the largest.
If that assumption is broken and the last layer is smaller than the
others, all layers that are bigger will be scaled up.

Bug: webrtc:10069
Change-Id: Iff87ddba741d5dfe3d0cc25a8f75d898a417eec7
Reviewed-on: https://webrtc-review.googlesource.com/c/112460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25878}
2018-12-03 14:07:14 +00:00
e7673cf922 RtcEventLogImpl nits
1. Make |output_period_ms_| optional, so as to clarify where
   it gets assigned a value. (I.e. the value set by the ctor
   is not retained.)
2. Some extra const modifiers.

Bug: webrtc:8111
Change-Id: I9f3ad7ff763cfbc9c9385f7fd4325ba696772765
Reviewed-on: https://webrtc-review.googlesource.com/c/112588
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25877}
2018-12-03 13:50:43 +00:00
aa3d8ad71b Reland "Fix output period in RtcEventLogImpl"
This is a reland of a4dcb749fbbc83a874d4e2c65de5a98465d3e200

Original change's description:
> Fix output period in RtcEventLogImpl
> 
> RtcEventLogImpl::StartLogging() was ignoring one of its parameters.
> This CL fixes the issue.
> 
> Bug: webrtc:10082
> Change-Id: Ie1790c1a7299748dabe99909d967384ad9895635
> Reviewed-on: https://webrtc-review.googlesource.com/c/112586
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25858}

Bug: webrtc:10082
Change-Id: I783fba84aa35e489f6235538c624b19f2f98a962
Reviewed-on: https://webrtc-review.googlesource.com/c/112860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25876}
2018-12-03 12:48:42 +00:00
ad82a424e4 Fix race over RtcEventLogImpl::task_queue_
RtcEventLogImpl::task_queue_ is a std::unique_ptr<rtc::TaskQueue>.
When a unique_ptr is destroyed, it first sets its internal pointer
to point to null, and only then invokes the destructor of that
object. However, the code in RtcEventLogImpl relies on
rtc::TaskQueue's property, that its destructor blocks on executing
tasks.

We solve by manually invoking the destructor, and only resetting
the internal pointer thereafter. In theory, we could have changed
the unique_ptr to a raw pointer at this point. We avoid that, so
as to keep the ownership clearer to readers of the code.

Bug: webrtc:10085
Change-Id: I54bbf5d6bae019757ca2e31ee960d558058ccc42
Reviewed-on: https://webrtc-review.googlesource.com/c/112598
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25875}
2018-12-03 12:41:21 +00:00
4da382e34d Speculative revert due to a long history of issues related to VP9 on Mac.
See https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc/WebRTC%20Chromium%20Mac%20Tester

First, we figured that "ba2840c Various VP9 high fps fixes by Ilya Nikolaevskiy" was the cause and it was reverted but it did not help.

We must now try the other CL which had done changed in VP9.

Revert "Reland Profile 2 to default profiles"

This reverts commit 4c0cc5bc5fa027b9392ff2886e731bea3aac7602.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Reland Profile 2 to default profiles
> 
> This is a reland after chrome browser tests are updated.
> 
> Bug: webrtc:9376
> Change-Id: I818bf5d447da7901ffe49f2c452decb89196e829
> TBR: niklas.enbom@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/c/112060
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25778}

TBR=emircan@webrtc.org

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

Bug: webrtc:9376
Change-Id: I3eb935c08341ce51fa16717ed7b3be5f5253aa2f
Reviewed-on: https://webrtc-review.googlesource.com/c/112597
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25874}
2018-12-03 12:06:10 +00:00
9cf6ce3e1b Finish renaming another tryjob that was forgotten
TBR: mbonadei@webrtc.org
No-Try: True
Bug: webrtc:10072
Change-Id: Ie9f53eabad9ab7f2bed92a38a0d640bd9315103d
Reviewed-on: https://webrtc-review.googlesource.com/c/112599
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25873}
2018-12-03 11:36:02 +00:00
11d3d6c17d Rename most tryjobs to avoid implied defaults
Explicitly say "_compile_" for ARM and Clang and GCC.
Explicitly say "_arm_" for mobiles.
Explicitly say "_x86_" for Windows.
Fill in some gaps where both tester and compile-only bots are viable.

Also remove unused "experimental" tryjobs.

No-Try: True
Bug: webrtc:10072
Change-Id: Ib22e0518fc1e600b237c3c687994f27c7e88b8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/112585
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25872}
2018-12-03 11:09:40 +00:00
3b923d95d5 Remove color space enum value kInvalid
kInvalid does not have a corresponding entry in the standard is therefore removed.
kUNSPECIFIED should be used instead.

Bug: webrtc:8651
Change-Id: Iee8cd85830aedaa4a9102251121b9975d40fa5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/112421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25871}
2018-12-03 09:53:02 +00:00
93dac8ec36 Delete workaround for CreateEvent
There used to be a collision between a macro in windows headers and
the CreateEvent method on EventFactory. But since the latter class is
deleted (see https://webrtc-review.googlesource.com/c/110140)
workaround no longer needed.

Bug: webrtc:3380
Change-Id: I4e2e3cfff4d7a99f7c22da289628839fdc5012b4
Reviewed-on: https://webrtc-review.googlesource.com/c/112593
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25870}
2018-12-03 09:23:22 +00:00