Commit Graph

29504 Commits

Author SHA1 Message Date
c2509fec7c Roll chromium_revision 2638d7649b..2bc032e864 (731908:732049)
Change log: 2638d7649b..2bc032e864
Full diff: 2638d7649b..2bc032e864

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b9bb682ff6..6a138fbe7c
DEPS diff: 2638d7649b..2bc032e864/DEPS

No update to Clang.

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

Change-Id: I2fc72b57cb14f1e1b32a3f969fea3c29a97a624e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166183
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@{#30272}
2020-01-15 18:45:40 +00:00
ea992f8771 Reland "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
This reverts commit d2d7a47247187236ce62e3c842963f6e4e9f0f1f.

Reason for revert: This revert is not needed. Failure was not due to webrtc.

Original change's description:
> Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
> 
> This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.
> 
> Reason for revert: Causing a build break:
> webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
> this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
>   'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'
> 
> 
> 
> Original change's description:
> > Reland "Extracts ssrc based feedback tracking from feedback adapter."
> > 
> > This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> > 
> > Original change's description:
> > > Extracts ssrc based feedback tracking from feedback adapter.
> > > 
> > > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > > 
> > > Bug: webrtc:9883
> > > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30076}
> > 
> > Bug: webrtc:9883
> > Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30266}
> 
> TBR=sprang@webrtc.org,srte@webrtc.org
> 
> Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9883
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30270}

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

Change-Id: Idd1073ebfef77b2154d7123b47dacb479537c550
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166200
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30271}
2020-01-15 18:24:32 +00:00
d2d7a47247 Revert "Reland "Extracts ssrc based feedback tracking from feedback adapter.""
This reverts commit d61338fa6ed957dd992f25da4844db34b14f89c7.

Reason for revert: Causing a build break:
webrtc/call/BUILD:300:1: Undeclared inclusion(s) in rule 'webrtc/call:rtp_sender':
this rule is missing dependency declarations for the following files included by 'call/rtp_transport_controller_send.cc':
  'webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h'



Original change's description:
> Reland "Extracts ssrc based feedback tracking from feedback adapter."
> 
> This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974
> 
> Original change's description:
> > Extracts ssrc based feedback tracking from feedback adapter.
> > 
> > This prepares for moving TransportFeedbackAdapter to TaskQueue.
> > 
> > Bug: webrtc:9883
> > Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30076}
> 
> Bug: webrtc:9883
> Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30266}

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

Change-Id: I7f3f872c7ff863a37ad8dca08051fe1e04671bfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166182
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30270}
2020-01-15 17:44:42 +00:00
219d8ce889 GOOG_PING: improve handshake
This patch improves handshake wrt GOOG_PING support so that
- if goog_ping_enable: sender send it's goog-ping version until it gets
STUN_BINDING_RESPONSE
- receiver only sends it's goog-ping-version if getting a
goog-ping-version in the request

This means that the overhead of STUN_ATTR_GOOG_MISC_INFO is only
- added on STUN_BINDING_REQUEST until a response is received.
- added on STUN_BINDING_RESPONSE if remote peer request it.

This is wire compatible with older versions so that
- new sender will enable GOOG_PING with new/old receiver.
- old sender will enable GOOG_PING with old receiver.
- old version will not enable GOOG_PING with new receiver
  (receiver expecting sender to announce first).

BUG: webrtc:11100
Change-Id: Ib3434c593988188150f4c7506918139aaf138d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165787
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30269}
2020-01-15 16:09:38 +00:00
a846cef197 Change rate stats classes to use int64_t not size_t
This avoids integer overflows when size_t is 32 bits, and conforms
to style guide recommendations to avoid unsigned integers.

Also add tests for overflow on RateStatistics accumulator.

Bug: webrtc:11247
Change-Id: Ifa0db567f41bbcf3ec46d89ab888f2ed9d03f3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163991
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30268}
2020-01-15 13:46:38 +00:00
7787ebcd3f Deflake CpuTimeTest.TwoThreads
The test sometimes failed because thread creation on some platforms on
internal tests may take too much work.

Now checks are less strict.

Bug: none
Change-Id: Ibd3df02bda26b0c5e804360a909c61afa760b3bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165960
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30267}
2020-01-15 13:36:58 +00:00
d61338fa6e Reland "Extracts ssrc based feedback tracking from feedback adapter."
This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974

Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
> 
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}

Bug: webrtc:9883
Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30266}
2020-01-15 12:51:16 +00:00
ccab06fb72 Revert "Replaces SynchronousMethodCall with rtc::Thread::Invoke."
This reverts commit b0e0728159f07269a875c5b53658603cf6733480.

Reason for revert:

Causes Chromium tests to timeout, preventing rolls into Chromium.

Original change's description:
> Replaces SynchronousMethodCall with rtc::Thread::Invoke.
> 
> Given that we already have Thread:.Invoke that can be used with lambda,
> SynchronousMethodCall doesn't add any value.
> 
> This simplification prepares for simulated time peer connection tests.
> 
> Bug: webrtc:11255
> Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30217}

TBR=steveanton@webrtc.org,srte@webrtc.org

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

Bug: webrtc:11255
Change-Id: I9d3aa218013129db7a09a77500a0547ce9ae341a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166047
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30265}
2020-01-15 12:34:35 +00:00
61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
07b17df771 Move DegradationPreference logic to the encoder queue.
This moves SetHasInputVideoAndDegradationPreference() to the encoder
queue. OveruseFrameDetectorResourceAdaptationModule is now entirely
single-threaded, including its inner class VideoSourceRestrictor.

VideoStreamEncoder now protects the module with RTC_GUARDED_BY. This
ensures it is safely used, even without a SequenceChecker inside of the
module. The module's |encoder_queue_| is removed.

The one task queue reference that is needed - passing down the current
task queue to StartCheckForOveruse() - is replaced by a TaskQueueBase*
(instead of rtc::TaskQueue*), enabling obtaining the current queue with
TaskQueueBase::Current(). (There is no rtc::TaskQueue::Current().)

Furthermore, the only uses of VideoSourceSinkController that isn't on
the encoder queue are documented, with a TODO saying if these are moved
the VideoSourceSinkController could also be made single-threaded.
However since this requires introducing a delay to
VideoStreamEncoder::SetSource() and VideoStreamEncoder::Stop(),
arguably a more risky change, if this is to be attempted that should be
in a separate CL.

Bug: webrtc:11222
Change-Id: I448ca5125708d5f66b95b0b180d6d24cc356dfa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165783
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30263}
2020-01-15 11:58:04 +00:00
6ef59d1ced Don't pace audio by default
After experimentation, not pacing audio is better. This is controlled
by the field trial WebRTC-Pacer-BlockAudio. This change keeps the flag,
but changes the behaviour such that it defaults to Disabled. However,
audio can still be paced if one chooses by enabling the field trial.

Bug: webrtc:11257
Change-Id: I5b23a82bb6708c007cf8dfb40065c821eefdc4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165381
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30262}
2020-01-15 11:21:14 +00:00
6dc06c8146 Roll chromium_revision 65afcfa031..2638d7649b (731779:731908)
Change log: 65afcfa031..2638d7649b
Full diff: 65afcfa031..2638d7649b

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7431e17d79..b9bb682ff6
* src/third_party/depot_tools: ce09ca54f8..2a04803267
DEPS diff: 65afcfa031..2638d7649b/DEPS

No update to Clang.

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

Change-Id: I8e2c541f42761099cafac182b3655ab81d5c1604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166121
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@{#30261}
2020-01-15 10:37:34 +00:00
d06588a758 Change Av1 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00
f1173f46e5 Revert "Using simulated rtc::Thread for peer connection scenario tests."
This reverts commit b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453.

Reason for revert: Interferes with other tests in same binary.

Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
> 
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}

TBR=steveanton@webrtc.org,srte@webrtc.org

Change-Id: If2e60edae264a4bb0dee3abf66ba2078fd85f493
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166045
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30259}
2020-01-15 10:10:07 +00:00
b70c5c5ce9 Using simulated rtc::Thread for peer connection scenario tests.
Bug: webrtc:11255
Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30258}
2020-01-15 09:35:40 +00:00
71574f7f3b Add data dependency to event_log_visualizer.
This .wav file is an implicit data dependency, this CL adds this
information to the build system.

Bug: None
Change-Id: Ia953e63d4658debce3cecb93bb1f3e749fe52f54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166044
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30257}
2020-01-15 09:21:10 +00:00
178a685ada Allow overwriting current thread in ThreadManager.
This prepares for introducing a simulated time rtc::ThreadManager
implementation that will run on a single underlying thread.

Bug: webrtc:11255
Change-Id: I793128cc0b8e649a3675914de67dfee3298b446a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165765
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30256}
2020-01-15 08:55:20 +00:00
21feefcf8e Roll chromium_revision c85b2ddbc8..65afcfa031 (731677:731779)
Change log: c85b2ddbc8..65afcfa031
Full diff: c85b2ddbc8..65afcfa031

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c8083d1681..7431e17d79
* src/third_party/depot_tools: 59a3b2fd5d..ce09ca54f8
DEPS diff: c85b2ddbc8..65afcfa031/DEPS

No update to Clang.

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

Change-Id: I445d3f44295a38a6e29e78bd71cac192ce26903a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166103
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@{#30255}
2020-01-15 02:43:34 +00:00
9921779540 Roll chromium_revision e61d470ddb..c85b2ddbc8 (731529:731677)
Change log: e61d470ddb..c85b2ddbc8
Full diff: e61d470ddb..c85b2ddbc8

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/13d8a54a70..c8083d1681
DEPS diff: e61d470ddb..c85b2ddbc8/DEPS

No update to Clang.

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

Change-Id: Id3db15313da66afaebd88de4c2f1b4cdb085f67f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166101
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@{#30254}
2020-01-14 22:36:48 +00:00
47b5d4cc5a Roll chromium_revision 81b1889c8c..e61d470ddb (731328:731529)
Change log: 81b1889c8c..e61d470ddb
Full diff: 81b1889c8c..e61d470ddb

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/25614ff33a..13d8a54a70
DEPS diff: 81b1889c8c..e61d470ddb/DEPS

No update to Clang.

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

Change-Id: Ic9dc736550213bbd67475faabf99d83791721589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166080
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@{#30253}
2020-01-14 18:44:48 +00:00
3d4d94a832 Adds scenario test for transport wide feedback based retransmission.
This ensures more end to end test coverage of the feature and captures
a wider class of regression then the existing unit test.

Bug: webrtc:9883
Change-Id: I6e74e571500c5c5d74caf8f661cac08bee8934f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164461
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30252}
2020-01-14 16:00:14 +00:00
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
6153e15d31 Roll chromium_revision a989226e28..81b1889c8c (731140:731328)
Change log: a989226e28..81b1889c8c
Full diff: a989226e28..81b1889c8c

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/49cfb9bdc2..25614ff33a
* src/third_party/depot_tools: a1266b63b5..59a3b2fd5d
* src/third_party/harfbuzz-ng/src: 64a45be519..82545c5e2b
DEPS diff: a989226e28..81b1889c8c/DEPS

No update to Clang.

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

Change-Id: I623a817f55cf7afa004d6b4b7f9ab16d7463d3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166020
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@{#30250}
2020-01-14 14:35:38 +00:00
ecc5b93b13 AEC3: Restrict default logging of some delay changes to VERBOSE
It leads to overly verbose test output. Example:
https://chromium-swarm.appspot.com/task?id=49bc386e0545ef10

Bug: webrtc:11278
Change-Id: I4a1c565f3aab94d98910722b23dcadc5fcde602a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165962
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30249}
2020-01-14 12:52:47 +00:00
3e66a498c3 Use RTX SSRCs in scenario test framework.
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
2020-01-14 12:04:56 +00:00
2a92d2b461 Cleanup: Prepares for simulated time peer connection tests.
This CL contains some preparatory cleanup that can be done
outside the main CL.

Bug: webrtc:11255
Change-Id: Ib0dcd81d352bafc446dcd2f7f82ba81f5e82e210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165766
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30247}
2020-01-14 09:55:42 +00:00
b580bff520 Roll chromium_revision d6f6958da9..a989226e28 (731013:731140)
Change log: d6f6958da9..a989226e28
Full diff: d6f6958da9..a989226e28

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/71813e2ccf..49cfb9bdc2
DEPS diff: d6f6958da9..a989226e28/DEPS

No update to Clang.

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

Change-Id: I9b42a9df1f3fba62cb529e3e98d986f0af194994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165940
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@{#30246}
2020-01-14 08:37:05 +00:00
1546f99572 Fixed timeout overflow in sctp reliability test.
Sometimes some tests failed due to test long execution while timeout
was computed to negative value.

Bug: None
Change-Id: Icb666170323f6b757a409db575d36116f57632d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165691
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30245}
2020-01-14 08:18:25 +00:00
3e3c551ac6 Suppress C5041 constexpr warning for MSVC 2019
Disable the C5041 warning which makes the build fail. This is a
C++17-only change and WebRTC doesn't support C++17 yet, so the code is
technically correct, but fails to build on MSVC 2019 and
warning-as-error active.

Also fix another warning-as-error build error with MSVC 2019 due to
ignoring the result of a [[nodiscard]] function.

No-Presubmit: True
Bug: webrtc:11275,webrtc:11276
Change-Id: I891a894ee87252f96e84fd8d282576f46907256f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30244}
2020-01-14 07:44:35 +00:00
2ea27968d3 Extract an interface from the perf results logger.
The new interface is called PerfTestResultWriter and is currently
implemented by PerfResultsLogger (renamed PerfTestGraphJsonWriter).

I plan to introduce a second implementation of the perf logger that
uses the new Histogram C++ API. I add a flag that chooses
between the two implementations so I can try it out (perhaps by
setting up a second, limited run of webrtc_perf_tests on the perf
bots that uses the new implementation). The histogram C++
implementation will come in the next patch.

As a side effect, I disentangled the plottable counter printer from
the perf result printer so it will work for both implementations.
The only thing they had in common was that both wrote JSON anyway.

See the bug for details on the new API.

Bug: chromium:1029452
Change-Id: Icb21b25ced08ea73aeecd221e9d51f2adf3dab1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165389
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30243}
2020-01-14 06:05:02 +00:00
145cfc5025 Roll chromium_revision 69c66e4366..d6f6958da9 (730870:731013)
Change log: 69c66e4366..d6f6958da9
Full diff: 69c66e4366..d6f6958da9

No dependencies changed.
No update to Clang.

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

Change-Id: I5b2bf79e71bc85ae0fe351a6365303462306e97f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165880
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@{#30242}
2020-01-14 02:37:17 +00:00
2cbfe17129 Roll chromium_revision 0792dc5faa..69c66e4366 (730752:730870)
Change log: 0792dc5faa..69c66e4366
Full diff: 0792dc5faa..69c66e4366

No dependencies changed.
No update to Clang.

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

Change-Id: I0187c15ad63120d2d88582137017a4b52d7e5f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165840
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@{#30241}
2020-01-13 22:44:43 +00:00
76294b6709 Roll chromium_revision 210e790756..0792dc5faa (730612:730752)
Change log: 210e790756..0792dc5faa
Full diff: 210e790756..0792dc5faa

Changed dependency
* src/third_party/depot_tools: 5e96ad12ac..a1266b63b5
DEPS diff: 210e790756..0792dc5faa/DEPS

No update to Clang.

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

Change-Id: Iab86c192804333fe7b7113224a6e5ce562f166ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165821
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@{#30240}
2020-01-13 20:40:08 +00:00
bb6677709b Roll chromium_revision b581de5b1b..210e790756 (730447:730612)
Change log: b581de5b1b..210e790756
Full diff: b581de5b1b..210e790756

Changed dependencies
* src/buildtools/linux64: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/mac: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/win: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/third_party/depot_tools: 7a8bf94894..5e96ad12ac
DEPS diff: b581de5b1b..210e790756/DEPS

No update to Clang.

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

Change-Id: I39e364b2aec00a902f8d665716c36e1fd48385da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165820
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@{#30239}
2020-01-13 18:47:00 +00:00
b8c775aeaf Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
Bug: webrtc:11251
Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-13 18:31:30 +00:00
8234b92ba3 Move DegradationPreference logic out of VideoSourceSinkController.
The DegradationPreference logic is moved into
OveruseFrameDetectorResourceAdaptationModule. This makes the adaptation
module solely responsible for degradation preference, and the
VideoStreamEncoder the only bridge between the adaptation module and the
VideoSourceSinkController.

The adaptation module is now unaware of the existence of a controller.
It only "speaks" VideoSourceRestrictions, which is a big milestone in
making adaptation modules injectable.

A follow-up CL will explore the possibility of reconfiguring the
controller's source and which degradation preference to use to the
encoder queue. This would allow us to make several classes
single-threaded, but it is a change in behavior and should be done in a
separate CL.

Bug: webrtc:11222
Change-Id: Ib7f640e12789da5f801177926c2072a51818f261
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165684
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30237}
2020-01-13 17:24:48 +00:00
e270ff1c41 [iOS] Reset VT session when H264 decoder malfunction error happen
Bug: webrtc:11268
Change-Id: I6932cfbe53dc7b922a90604de799f259526b4c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165785
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30236}
2020-01-13 14:57:36 +00:00
6ea2c6ae87 Cleanup: Merges Thread and MessageQueue.
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.

Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
2020-01-13 13:53:20 +00:00
7d43801a07 Delete RtpGenericDepacketizer as no longer used
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
48148dc840 Change log level of AEC3 buffer info to VERBOSE
Otherwise, test logs become very verbose:
https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
See linked issue.

Bug: webrtc:11278
Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30233}
2020-01-13 13:31:27 +00:00
fa73393574 In TaskQueueWin fix race in canceling MutlimediaTimer
Bug: webrtc:11232
Change-Id: I371f0b78a572c94f2eefd8e0859eed88bce9e37e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165762
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30232}
2020-01-13 13:02:16 +00:00
fae640003c Add saza@ and peah@ to OWNERS of some audio files
Bug: None
Change-Id: Ibab0528b09bf2c4f0af4fd383a7b5e93e6c55f6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165784
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30231}
2020-01-13 12:31:21 +00:00
54891af88d More lenient double comparison for RunningStatistics.FullSimpleTest
The test was failing (in a flaky fashion, interestingly), comparing:
* 50.5 whose mantissa is:
1001010000000000000000000000000000000000000000000000
* with 50.500000000000036 whose mantissa is:
1001010000000000000000000000000000000000000000000101

since EXPECT_DOUBLE_EQ() only allows 4 ULPs difference.
We don't need this kind of precision.

Bug: webrtc:11134
Change-Id: I811178b0762dbcd61d4f2d3f047ea0b59847fa57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165761
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30230}
2020-01-13 12:21:12 +00:00
53cd9e2645 Separates simulated TaskQueue and simulated ProcessThread.
The overlap in functionality is quite limited and separating the
functionality makes it a bit easier to follow each. This prepares
for adding a SimulatedThread class in a follow up CL.

Bug: webrtc:11255
Change-Id: I83c754bd570113dfb582098bb4d39e27bb4f4d87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165688
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30229}
2020-01-13 11:26:42 +00:00
ce0ea49001 VideoStreamEncoder configuring source/sink with VideoSourceController.
This is part of the work for making VideoStreamEncoder responsible for
configuring its source/sink and limiting the responsibility of
OveruseFrameDetectorResourceAdaptationModule to only output relevant
VideoSourceRestrictions.

BEFORE THIS CL

Prior to this CL, OveruseFrameDetector was responsible for performing
AddOrUpdateSink() on the source, which it did using its nested class
VideoSourceProxy.

AddOrUpdateSink() could happen for both adaptation and non-adaptation
related reasons. For example:
- Adaptation related: AdaptUp() or AdaptDown() happens, causing updated
  VideoSourceRestrictions.
- Non-adaptation related: VideoStreamEncoder asks the module to
  reconfigure the source/sink for it, such as with
  SetMaxFramerateAndAlignment() or SetWantsRotationApplied().

AFTER THIS CL

AddOrUpdateSink() is performed by VideoSourceController, which is owned
by VideoStreamEncoder. Any reconfiguration has to go through the
VideoStreamEncoder. This means that:
- Non-adaptation related settings happen between VideoStreamEncoder and
  VideoSourceController directly (without going through the adaptation
  module).
- Adaptation related changes can be expressed in terms of
  VideoSourceRestrictions. OveruseFrameDetectorResourceAdaptationModule
  only has to output the restrictions and not know or care about other
  source/sink settings.

For now, VideoSourceController has to know about DegradationPreference.
In a future CL, the DegradationPreference logic should move back to
the adaptation module. The VideoSourceRestrictions are fully capable of
expressing all possible source/sink values without the "modifier" that
is the degradation preference.

Bug: webrtc:11222
Change-Id: I0f058c4700ca108e2d9f212e38b61f6f728aa419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162802
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30228}
2020-01-13 11:14:04 +00:00
d238200882 Introduce ResourceAdaptationModuleListener and VideoSourceRestrictions.
The VideoSourceRestrictions describe the maximum pixels per frame and
max frame rate of a video source.

This CL makes the ResourceAdaptationModuleInterface responsible for the
reconfiguration of video sources. The VideoSourceRestrictions is the
output of an adaptation module, and the ResourceAdaptationModuleListener
handles the callback for when the source restrictions change.

The OveruseFrameDetectorResourceAdaptationModule is updated to output
its changes using these interfaces, and VideoStreamEncoder - now a
listener - is made responsible for triggering the reconfiguring the
video source.

Performing the reconfiguration still requires interacting with the
VideoSourceProxy - it is still partially responsible for keeping track
of rtc::VideoSinkWants settings and performing AddOrUpdateSink(). For
now this may look a bit weird: the VideoSourceProxy tells the
VideoStreamEncoder about the new restrictions, and then the
VideoStreamEncoder tells the VideoSourceProxy to apply these
restrictions on the source/sink. This exercises the listener though, and
unblocks the next CL.

The next CL should move all "configuring the source" logic to the
VideoStreamEncoder instead, so that the only information that is tracked
by OveruseFrameDetectorResourceAdaptationModule is what it actually
outputs to the listener. See the next CL
(https://webrtc-review.googlesource.com/c/src/+/162802) where a
VideoSourceController is introduced, to be owned by the
VideoStreamEncoder rather than the adaptation module.

Bug: webrtc:11222
Change-Id: I450ce74f51d96c4b98009a06134db671893d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162522
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30227}
2020-01-13 10:57:00 +00:00
2869638b4d Avoid [[nodiscard]] warning C4834 with MSVC 2019
Avoid a warning-as-error of MSVC 2019 due to a test ignoring a
[[nodiscard]] return value:

C4834: discarding return value of function with 'nodiscard' attribute
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d

Bug: webrtc:11275
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164467
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30226}
2020-01-13 10:07:35 +00:00
f5ecb5f22e Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""
This reverts commit 9cad4dccc96f09d76dce3766a076052df7d75ab8.

Reason for revert: Breaks downstream tests.

Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive video codecs"""
> 
> This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c
> 
> This CL lands all code except the code that activates the change,
> see media/engine/webrtc_video_engine.cc
> Once downstream projects are fixed, there will be a one-line change to
> activate the change to distinguish between send and receive video codecs.
> 
> Original change's description:
> > Reland "Reland "Distinguish between send and receive video codecs""
> >
> > This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> > >
> > > Reason for revert: Downstream test updated.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive video codecs""
> > > >
> > > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > > >
> > > > Reason for revert: Breaks another downstream test.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > > >
> > > > > Reason for revert: Downstream tests have been updated.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive video codecs"
> > > > > >
> > > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > > >
> > > > > > Reason for revert: Breaks downstream test.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive video codecs
> > > > > > >
> > > > > > > Even though send and receive codecs are the same,
> > > > > > > they might have different support in HW.
> > > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > > track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30079}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30097}
> >
> > Bug: chromium:1029737
> > Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30120}
> 
> Bug: chromium:1029737
> Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30219}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

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

Bug: chromium:1029737
Change-Id: I377f82866e56862f57383f96a3b96719344eef9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30225}
2020-01-13 09:03:37 +00:00
081f7a3498 Add missing header for dchecked_cast on UWP
Add missing #include to fix some build error on `winuwp` with some code
using rtc::dchecked_cast<> under an `#ifdef (WINUWP)`, resulting in an
undefined symbol error.

Bug: webrtc:11194
Change-Id: Iad9e74c3e92ed6cf1461f34cdd9329d13f5d62e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161721
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30224}
2020-01-13 07:53:49 +00:00
41d96d26ee Flip goog_ping_announce default to false
BUG: webrtc:11100
Change-Id: I37a23b32b339c000cc2e88793c31732f7f1d259d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165686
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30223}
2020-01-13 06:38:35 +00:00