Commit Graph

26553 Commits

Author SHA1 Message Date
d841ea6b58 Skip return value check for stopped repeating tasks.
If the repeated task stopped itself, the return value does not matter,
therefore we don't have to check it.

Bug: webrtc:10365
Change-Id: I4a2c7a40a287dd0a8099628e228e9c319409576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127545
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27121}
2019-03-14 08:31:19 +00:00
5d7e43936e Roll chromium_revision acb568e104..fc637deb51 (640514:640618)
Change log: acb568e104..fc637deb51
Full diff: acb568e104..fc637deb51

Changed dependencies
* src/base: 134665857c..584001face
* src/build: 787c9a46a5..2678ddc6fc
* src/buildtools: 672ce635e9..44579472d1
* src/ios: cdcf84d83d..4a091ba968
* src/testing: df90689544..508791909a
* src/third_party: 0d5c714625..86d240affe
* src/third_party/depot_tools: deb384f985..40c19421b4
* src/tools: 627d0f41c5..61b1f4bc90
DEPS diff: acb568e104..fc637deb51/DEPS

No update to Clang.

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

Change-Id: I607820b7bdddb4383f148a2f5aa275f9217def3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127701
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@{#27120}
2019-03-14 06:31:14 +00:00
3cc45d4467 Add a test that all //api/test headers are compilable.
Bug: webrtc:10376
Change-Id: I2a1ea24ddf5980c76660724fae68c16179bb25a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125682
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27119}
2019-03-14 05:27:53 +00:00
d5e1c372c9 SSLCertificate basic fuzzer.
This change simply calls through all code paths in the SSLCertificate interface
after passing in an untrusted PEM string. Corpus will follow in another CL.

Bug: webrtc:10395
Change-Id: I001642fa89a84ce01505780f5e76f01a0e46a785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127640
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27118}
2019-03-14 03:53:24 +00:00
3aa584ff11 Fixes ClangTidy issues in api/
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: I20190c01559ff315422be1b3f980853cbc5afbcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127625
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27117}
2019-03-14 03:52:19 +00:00
ce66bb4d81 Adding simulcast examples to the fuzzing corpus.
Adding an example of a request to send simulcast (from the PC).
Adding an example of a request to receive simulcast (from the SFU).

Bug: webrtc:10409
Change-Id: I13b689621e2f89f8e00b7ee8bc542157ccebb873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127621
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27116}
2019-03-14 01:10:08 +00:00
1295b0def0 Add basic fuzzing for rtp_header_parser.h/cc.
rtp_header_parser currently has 0% fuzzing coverage. To improve this I have
added a basic fuzzer which fuzzes all of the available paths.

Bug: webrtc:10395
Change-Id: I30324b2bfa7629b0110527258b33b7e048e89fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27115}
2019-03-13 23:31:16 +00:00
ec4cdbadc2 Roll chromium_revision a9ac2956aa..acb568e104 (640406:640514)
Change log: a9ac2956aa..acb568e104
Full diff: a9ac2956aa..acb568e104

Changed dependencies
* src/base: 742fb6ccae..134665857c
* src/build: 2a1991f5f2..787c9a46a5
* src/buildtools: 794f2d1f1c..672ce635e9
* src/ios: c567f8dcf8..cdcf84d83d
* src/testing: f042685793..df90689544
* src/third_party: 85e6a62d44..0d5c714625
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/293a9a2e27..2afe880da0
* src/third_party/depot_tools: 8b94108e68..deb384f985
* src/tools: 60ab88469b..627d0f41c5
DEPS diff: a9ac2956aa..acb568e104/DEPS

No update to Clang.

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

Change-Id: I126206280041ee7407c08746c225027a6ce92b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127622
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@{#27114}
2019-03-13 23:26:06 +00:00
7f3687ce26 Integrate parsing of SCTP messages into WebRTC Fuzzers.
This change adds a basic fuzzer to exercise parsing of SCTP messages.

Bug: webrtc:10395
Change-Id: I1fd7a8560add3463c1978ebcad30082ae31f2073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27113}
2019-03-13 20:52:46 +00:00
7a7cf94a04 Roll chromium_revision c0acb51236..a9ac2956aa (640306:640406)
Change log: c0acb51236..a9ac2956aa
Full diff: c0acb51236..a9ac2956aa

Changed dependencies
* src/base: b5856ccecc..742fb6ccae
* src/build: c8385dae5f..2a1991f5f2
* src/ios: 509dd1aaa3..c567f8dcf8
* src/testing: 4dbdb9e4e1..f042685793
* src/third_party: 221c01a24e..85e6a62d44
* src/tools: 8c81fdf25c..60ab88469b
DEPS diff: c0acb51236..a9ac2956aa/DEPS

No update to Clang.

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

Change-Id: I0cf7e77f78fe2547694fe23fad10a85a2085c42c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127604
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@{#27112}
2019-03-13 19:01:46 +00:00
45a2cd2b32 Fixing documentation for CopyOnWriteBuffer.
The copy constructor mistakenly claims that data is copied.
This change corrects the documentation to indicate that the
buffer data will be shared between the two objects.

Bug: None
Change-Id: Ie0d513bfd2f4de660d60c46e87afbf02f0ea3991
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127343
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27111}
2019-03-13 17:50:16 +00:00
d6c4b80268 Add Fuzzing support for ParseRtcpPacketSenderSsrc.
This function is called on each incoming RTCP payload.

Bug: webrtc:10395
Change-Id: I164746fe45912cc503565e77046b5d884e0204e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127122
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27110}
2019-03-13 17:28:56 +00:00
baf271f978 DefaultVideoQualityAnalyzer cleanup.
Remove done todos

Bug: webrtc:10138
Change-Id: I33ad6da41bf51a0ed3bafd5ae671f94d00e16ce3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127563
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27109}
2019-03-13 16:58:28 +00:00
982dc791db Preserve legacy behavior for old OveruseFrameDetector
A recent refactoring introduced a subtle difference in how encoded
frame timing is measured. See
https://webrtc-review.googlesource.com/c/src/+/124122/14/video/video_stream_encoder.cc#b1278

After that change, the encode was considered done after encoding was
done. The old behavior included the time needed to call stats and
the video sink, which might include video quality tests related tasks
and RTP packetization.

In order to preserve the old behavior I'm moving timestamping to after
packetization again.
Note that the timing frame info still has a separate timestamp that
does explicitly measure encode time. This is used by the experimental
new overuse detector, so the effect of this change will be transient
anyhow.

Bug: chromium:941457, webrtc:10164
Change-Id: Ia990a1ceaeaf2c45d5df2a32d4f017cdb08e3c55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127569
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27108}
2019-03-13 16:42:38 +00:00
5ce38fff17 Making UpdatesTargetRateBasedOnLinkCapacity more robust.
This CL adds enough simulated time to recover the built up delay. This
makes the test less sensitive to small timing changes. This prepares
for further changes in Scenario test framework.

Bug: webrtc:10365
Change-Id: Iddbe6a57e31f17f590004e29221f907321cbb3d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127567
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27107}
2019-03-13 15:36:55 +00:00
5ad789ceff Reland "NetEQ RTP Play: Optionally write output audio file"
This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a.

Reason for revert: issue fixed

Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
2019-03-13 15:33:29 +00:00
123f3453e2 Cleanup of scenario test framework.
* Removing unused return values.
* Using TaskQueueForTest to do blocking calls.
* Improving naming.

This prepares for future work to run scenario tests in simulated time.

Bug: webrtc:10365
Change-Id: I2c100e9c20f4b85e85d7b455ea01944f6a14e08f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127561
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27105}
2019-03-13 15:09:28 +00:00
9a66d5ed65 Add support to audioproc_f to generate a custom call order file.
This adds a flag to audioproc_f to generate a custom call order
file from an AEC dump. This file can be used to get more realism
when simulating with wav-files.

Bug: webrtc:10393
Change-Id: I245533d18affaab2f6cef53138332d7d83c71822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126782
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27104}
2019-03-13 15:08:18 +00:00
f84b95dbec Rename network_manager -> emulation.
Rename network_manager -> network_emulation_manager in the
network_emulation_pc_unittest.cc

Bug: webrtc:10138
Change-Id: I5df29f22d3d570bce1701d43d54d6d40f703b19b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127523
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27103}
2019-03-13 15:03:32 +00:00
3c589beee6 Reland "Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video."
This is a reland of 184f6d5d75c198cb7b70b8f9b75e0b5096c6e577.

Incorrect build dependencies in downstream tests have been fixed,
and an initialization bug in this CL has also been fixed.

Original change's description:
> Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video.
>
> This allows external users of this test fixture to specify a custom
> path, rather than just a custom file name.
>
> Bug: webrtc:10349
> Change-Id: I84e886c8bc28583017ce9ed7b9e7ee6a8e95730f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126227
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27033}

TBR: kwiberg@webrtc.org
Bug: webrtc:10349
Change-Id: I0ec9dd26cd96c3db8ac8482893a26e62a1b1eefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127181
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27102}
2019-03-13 15:00:05 +00:00
c032109cda Improve bitstream dumping logic in VideoReceiveStream
Specifying relative path like "~" somehow doesn't work in some cases.
To pass an absolute path some parameter rewriting is required.
Also, as the stream gets recreated several times at the beginning of the
call, append the time to the filename to make them unique.

Bug: none
Change-Id: I1c914f9081adb4ac5c34584d96e542e8a863547b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27101}
2019-03-13 14:59:02 +00:00
133b307c7d Delete method VideoStreamDecoder::UpdateRtt
It's routed to the old jitter buffer, which is instantiated but
unused.

Bug: webrtc:7408
Change-Id: I8fe2d8a2bfa6c2119945cc294d17982af4f830cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127541
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27100}
2019-03-13 14:57:20 +00:00
ecc11eb6e7 Roll chromium_revision 48038209dc..c0acb51236 (640007:640306)
Change log: 48038209dc..c0acb51236
Full diff: 48038209dc..c0acb51236

Changed dependencies
* src/base: 351141a2c6..b5856ccecc
* src/build: 477cc2028c..c8385dae5f
* src/ios: 0a2c8ecd8b..509dd1aaa3
* src/testing: bdb01bb642..4dbdb9e4e1
* src/third_party: 3a944a1807..221c01a24e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a7314e4d23..293a9a2e27
* src/third_party/depot_tools: 77e5d48a08..8b94108e68
* src/tools: 1eb8f742ae..8c81fdf25c
DEPS diff: 48038209dc..c0acb51236/DEPS

No update to Clang.

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

Change-Id: I01bcc2b2616a1aac5510a4ecb5a024770092abb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127600
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@{#27099}
2019-03-13 14:40:48 +00:00
cc3503248f Replace abs with std::abs in audio_coding/neteq/histogram.cc
Bug: None
Change-Id: Ie3277558aa27dd76e06ec1fc3cb49cdcf3d982d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125703
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27098}
2019-03-13 14:18:18 +00:00
208634763a Move creation of rtc::NetworkManager into network emulation layer
Bug: webrtc:10138
Change-Id: I64271fab46a8dccb09f255eb14a4404b0bccdea3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127285
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27097}
2019-03-13 12:52:49 +00:00
9699f09eb3 Add new webrtc_perf_test for lower stream of vp8 simulcast screenshare
Bug: none
Change-Id: Ic5b20b09449d7e36f638cc9f46beaa1f4099f98e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127287
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27096}
2019-03-13 10:18:54 +00:00
1175ae0d80 Add log based GoogCC simulation to visualizer.
This CL adds  a mode to simulate roughly what GoogCC could have been
doing during the recording of an rtc event log by using the logged
events as input to GoogCC and visualizing the resulting target rate.

This is similar to the existing simulated_sendside_bwe mode, but uses
the new NetworkControllerInterface to ensure more reliable GoogCC
simulation.

Bug: None
Change-Id: I57894aa666151efc8405407d928b5257fb9b7d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123924
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27095}
2019-03-13 09:39:14 +00:00
7ae8d64dd0 Restore VideoCodecInitializer to use only the 1st stream maxFramerate
This is a partial fix for regression introduced in
https://webrtc-review.googlesource.com/c/src/+/125461

Currently, the OveruseFrameDetector::OnTargetFramerateUpdated is called
only then the encoder is reconfigured, with the default maxFramerate.

Changing it from default 5 to 60, or even 30 made the detector too
sensitive and it caused adaptation down due to CPU overuse even on
powerful machines.

Bug: webrtc:10310, chromium:940466
Change-Id: I7b0eabfc8f9b502e293af1a5b02fc5d4ab468c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27094}
2019-03-13 09:07:47 +00:00
77efcd82db Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule."
This is a reland of 1b871d07532c25d2f27e4db192cb9ce2229b1cee

Original change's description:
> Replacing rtc::Thread with task queue for TestAudioDeviceModule.
>
> This prepares for running it in simulated time.
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:10465
> Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27083}

TBR=henrika@webrtc.org

Bug: webrtc:10465
Change-Id: Icda8043fb5b1156129bc3b706bf8f190782b0921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27093}
2019-03-13 09:01:05 +00:00
793597b7c9 Removes TaskQueueBase::Current call in repeating task.
Bug: webrtc:10365
Change-Id: Ie9fd1a28e96680b8b4b6ef8339c3eaba21ef9e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27092}
2019-03-13 08:17:22 +00:00
cda86dd483 Removes usages of repeating task without task queue argument.
This prepares from removing the overload in a followup CL.

Bug: webrtc:10365
Change-Id: I80db16e7d37944e3dc7d2799bbf45ef8f439a22c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126860
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27091}
2019-03-13 08:07:56 +00:00
e7a5f7bfae Modifying MediaChannel to accept CopyOnWriteBuffer by value.
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.

Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
2019-03-12 23:49:57 +00:00
dfaea9dd98 Fuzz rtc::StringToNumber.
StringToNumber is directly used in parsing the SDP so it should be fuzzed.

Bug: webrtc:10395
Change-Id: I85b520fbefd34d3dba49950c5ff297b482c572b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127123
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27089}
2019-03-12 22:05:46 +00:00
6a5e976fbe Add generic depacketizer fuzzer to WebRTC.
The generic video depacketizer was missed in the initial fuzzing pass.

Bug: webrtc:10395
Change-Id: I166f27fc5897a2eafe38dad8e074834fefcc330e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127041
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27088}
2019-03-12 20:48:19 +00:00
ade5cb8294 Field trial fuzzer.
This simple fuzzer is intended to detect potential issues in the field trial
parsing code. Since these can be set by the browser it is better to have some
fuzzing coverage around this area.

Bug: webrtc:10395
Change-Id: I1b8b859d2107a0bc99cb7520cf0ef96f3d110547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127121
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27087}
2019-03-12 20:47:15 +00:00
fa852efb73 Revert "Replacing rtc::Thread with task queue for TestAudioDeviceModule."
This reverts commit 1b871d07532c25d2f27e4db192cb9ce2229b1cee.

Reason for revert: Breaks webrtc downstream projects.

Original change's description:
> Replacing rtc::Thread with task queue for TestAudioDeviceModule.
> 
> This prepares for running it in simulated time.
> 
> TBR=henrika@webrtc.org
> 
> Bug: webrtc:10465
> Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27083}

TBR=henrika@webrtc.org,ossu@webrtc.org,srte@webrtc.org

Change-Id: I16d7c2a46d38c9aaf82cc3ab7bd7b9c5e10f5a5e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127341
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27086}
2019-03-12 20:05:53 +00:00
bd50a84336 Revert "Reland "DCHECK feedback_rtt is positive""
This reverts commit ab0d03dcaaa98677348d15352f7bc1e8e066180f.

Reason for revert: Broke webrtc importer.

Original change's description:
> Reland "DCHECK feedback_rtt is positive"
> 
> This is a reland of 37d4f91db3fd1d0e023a820d0cda86bb786a0f72
> Reason for reland: Got Aliby for Android FEC test flakes.
> 
> Original change's description:
> > DCHECK feedback_rtt is positive
> >
> > Bug: None
> > Change-Id: I6eb10d6a20a679fff08f604441f8e58dcd417608
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126464
> > Commit-Queue: Evan Shrubsole <eshr@google.com>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27034}
> 
> Bug: None
> Change-Id: Iacacda9e3b141c69189f7931a1ec63d74b2dd845
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126920
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27082}

TBR=srte@webrtc.org,yinwa@webrtc.org,yvesg@webrtc.org,yvesg@google.com,eshr@google.com

Change-Id: Ia59f20019309c1e0b44029179d63558e92d39a85
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127324
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27085}
2019-03-12 19:40:25 +00:00
ea07650b0b Roll chromium_revision 8158f07c24..48038209dc (639862:640007)
Change log: 8158f07c24..48038209dc
Full diff: 8158f07c24..48038209dc

Changed dependencies
* src/base: 1b069d3e88..351141a2c6
* src/build: ac2f1f6da8..477cc2028c
* src/ios: 2617208b4d..0a2c8ecd8b
* src/testing: d9c8fc7872..bdb01bb642
* src/third_party: bc7297d100..3a944a1807
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/eb73833a29..a7314e4d23
* src/tools: 848a110b95..1eb8f742ae
DEPS diff: 8158f07c24..48038209dc/DEPS

No update to Clang.

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

Change-Id: If4b974f4b6cee911dc56adeac68cfd299d94bc28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127323
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@{#27084}
2019-03-12 19:04:54 +00:00
1b871d0753 Replacing rtc::Thread with task queue for TestAudioDeviceModule.
This prepares for running it in simulated time.

TBR=henrika@webrtc.org

Bug: webrtc:10465
Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27083}
2019-03-12 18:25:31 +00:00
ab0d03dcaa Reland "DCHECK feedback_rtt is positive"
This is a reland of 37d4f91db3fd1d0e023a820d0cda86bb786a0f72
Reason for reland: Got Aliby for Android FEC test flakes.

Original change's description:
> DCHECK feedback_rtt is positive
>
> Bug: None
> Change-Id: I6eb10d6a20a679fff08f604441f8e58dcd417608
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126464
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27034}

Bug: None
Change-Id: Iacacda9e3b141c69189f7931a1ec63d74b2dd845
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27082}
2019-03-12 17:00:51 +00:00
ec65e1fee5 Allow construction of TaskQueueForTest with TaskQueueBase
Using simulated time implementation of TaskQueue with TaskQueueForTest.

Bug: webrtc:10365
Change-Id: I9f853133aa9fcd5858a5e2e1e0f47ab1a6a5ca18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127292
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27081}
2019-03-12 16:31:39 +00:00
1b4254ada5 Check current buffer time span instead of number of samples in postpone decoding after expand.
Bug: webrtc:10392
Change-Id: I2ad4d8c7a3f87cab32e2ea097b2e05aa179e0bc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126761
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27080}
2019-03-12 16:25:24 +00:00
075e7fdd00 Delete VCMPacket constructor with WebRtcRTPHeader
Bug: webrtc:10397
Change-Id: I8f51616d6953af6b2d3bdfdf46b748b2a78702ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127240
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27079}
2019-03-12 15:18:41 +00:00
c4b391a257 Revert "NetEQ RTP Play: Optionally write output audio file"
This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f.

Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.

Original change's description:
> NetEQ RTP Play: Optionally write output audio file
> 
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
> 
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
> 
> This CL also includes a test script with which the retro-compatibility
> has been verified.
> 
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
2019-03-12 14:28:23 +00:00
e096004745 Enable configuring probes via field trial.
This CL adds a field trial that lets us control the size of the initial probes, how we grow the following probes and how big and frequent our ALR probes are.

Bug: webrtc:10394
Change-Id: I6c7783dfada9aaf55cd836dd8991bb7b8ca4993b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27077}
2019-03-12 14:27:18 +00:00
cb96809e46 Make FieldTrialOptionals operator bool() explicit
Implicit bool conversions behave wierdly in a bunch of cases, so let's make it explicit.

Bug: None
Change-Id: I15933e90d57c57218eed9608407aace5a640a6ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127284
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27076}
2019-03-12 14:24:48 +00:00
bf40c380ae Pass flexfec_sender only to the protected media send stream.
Fixes a regression from cl
https://webrtc-review.googlesource.com/c/src/+/123187.

Bug: webrtc:7135
Change-Id: Ifdbb3a3fd61610ac2b4d1f454aef71791f2a5b04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127286
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27075}
2019-03-12 14:18:08 +00:00
fc6ab00a39 Introduce EmulatedRoute
Introduce a handle for route created with network emulation layer,
that can be used to remove it in future properly.

Bug: webrtc:10138
Change-Id: I9fb847caeee24333bafb328727711af005b09224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127283
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27074}
2019-03-12 13:25:32 +00:00
a268b69037 Rename EndpointConfig into EmulatedEndpointConfig
Also fix minor issues in this class.

Bug: webrtc:10138
Change-Id: Icb3ec7f6296c34da260e701ec51d7b87ce62a4d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127281
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27073}
2019-03-12 13:24:29 +00:00
30e60d6fd9 Remove dependency on DirectShow baseclasses (streams.h from the winsdk_samples directory).
This required a fairly extensive overhaul:

* Removed locks from the implementation.
  The filter/pin architecture does use multiple threads,
  but the state is controlled on one and synchronization
  is done via flags that don't require locking.
  Note though that the baseclasses used a lot of locking, it's unclear why,
  but perhaps there are things I'm not aware of. The locking was not done
  consistently though, which doesn't seem to have been a problem.
* Change the code to not mix AddRef/Release and use of explicit 'delete'.
* Removed implementations of interfaces we don't need/use.
* Similarly some methods now return E_NOTIMPL.
* Added some utilities to make use of COM interfaces and concepts, easier.

BUG=webrtc:10374
TBR=mbonadei@webrtc.org

Change-Id: Iaedb1157d37ef5d5c75f727dba3d7de75ce22cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125086
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27072}
2019-03-12 11:52:20 +00:00