Commit Graph

25094 Commits

Author SHA1 Message Date
80cd25bcfb Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
Bug: None
Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107303
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25312}
2018-10-23 12:13:02 +00:00
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
8ee06a7b0c Increase coverage of AEC3 JSON config unit tests, fix bugs
The new test checks that json strings are unchanged when parsing to a
config and back to a string. This ensures that everything in the json
representations is parsed when created a config from the json.

This CL also adds the render_levels config substruct to the JSON parser.

Some issues were surfaced by the new test:
 - Config validation clamping silently passed NaNs
 - Config validation only fixed the first out-of-bounds parameter, and
   not any subsequent ones
 - Config validation did not check all values in the config

Bug: webrtc:9535
Change-Id: Ie7b588731dc1fe26ba71d1eb2f177f3b3b8139e3
Reviewed-on: https://webrtc-review.googlesource.com/c/107120
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25310}
2018-10-23 11:08:54 +00:00
f7a7c8a8bc Stop adding RTT delay if there was not packet loss for enough time
Bug: none
Change-Id: I9105c0ee6a4f75381e133a496c3728dece9aca4b
Reviewed-on: https://webrtc-review.googlesource.com/c/23261
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25309}
2018-10-23 10:38:39 +00:00
0627e21a1a Removes unused DeliverPacket from CallClient.
Bug: webrtc:9510
Change-Id: Idfdce13ef407449c3896ad400ab4b8fb3ef589a1
Reviewed-on: https://webrtc-review.googlesource.com/c/107420
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25308}
2018-10-23 09:30:46 +00:00
9581bc4c52 Rename too long variable name to extmap_allow_mixed
Bug: webrtc:7990
Change-Id: I990111e473553163cecb9c73fec90d07c24aca02
Reviewed-on: https://webrtc-review.googlesource.com/c/107362
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25307}
2018-10-23 09:25:19 +00:00
2edab4c026 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-23 09:24:15 +00:00
01cf44d397 AEC3: Adding missing elements to the json parser
Bug: webrtc:9894
Change-Id: I10c23c5df5b262cfb07fc89715e12bc99fc7be3f
Reviewed-on: https://webrtc-review.googlesource.com/c/107201
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25305}
2018-10-23 09:23:11 +00:00
3583693638 3 TLs: add full stack test for short pattern + base heavy alloc.
A previous CL split an existing field trial in two. This CL
ensures that we have perf coverage for the two interesting
cases: short pattern and short pattern + base heavy TL alloc.

Tested: autoninja -C out/Release; and out/Release/webrtc_perf_tests --gtest_filter="*3TL*"
Bug: chromium:893500
Change-Id: I0585d67860d8a10122793fa1336440c13ebd0c57
Reviewed-on: https://webrtc-review.googlesource.com/c/107561
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25304}
2018-10-23 09:09:21 +00:00
6ed37ba1b5 AEC3: Enable fuzzer testing of old render buffering code.
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.

Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
2018-10-23 08:43:00 +00:00
d34597cf3d Update test::CreateVideoStreams to use num_temporal_layers.
MultiCodecReceiveTest/VideoSendStreamTest:
Configure num_temporal_layers via VideoEncoderConfig (and remove
implementations of VideoStreamFactoryInterface used to override
the default num_temporal_layers configuration).

Bug: none
Change-Id: I9855245477fe3c6fe48d1a755d401d6a35a17c70
Reviewed-on: https://webrtc-review.googlesource.com/c/107301
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25302}
2018-10-23 08:36:25 +00:00
98f5f6cdea In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,

new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.

Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.

Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
2018-10-23 08:29:34 +00:00
b0ab2ce256 Reland "Remove the HighPassFilter interface"
Downstream Chromium dependencies fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/1286449

This is a reland of e2405c1a823f3baf90a9c72f2e058f91eb659c20

Original change's description:
> Remove the HighPassFilter interface
>
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
>
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

Bug: webrtc:9535
Change-Id: I0017193ad3ca1762e186f3ad79f29d33ef468202
Reviewed-on: https://webrtc-review.googlesource.com/c/106681
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25300}
2018-10-23 07:44:09 +00:00
c9f9b8711f AEC3: Improve dominant nearend detection
This change makes the dominant nearend detection more accurate.
- The hangover is increased not leave nearend state between words.
- The SNR requirement is increased to not enter nearend state without
  speech activity.
- An early exit mechanism has been added to leave nearend state quickly
  when the echo is strong.

Bug: chromium:897701,webrtc:9897
Change-Id: I9e0f3e6ecb80eee1c0c917d4835f110555f74acf
Reviewed-on: https://webrtc-review.googlesource.com/c/107347
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25299}
2018-10-23 07:05:46 +00:00
ac19414512 Export symbols needed by the Chromium component build (part 6).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5
Reviewed-on: https://webrtc-review.googlesource.com/c/107400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25298}
2018-10-23 06:48:51 +00:00
9a5da497b5 Split out a separate target for SimulcastEncoderAdapter
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.

Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-22 22:36:59 +00:00
165148d84d Reland "Remove deprecated barcode scanning functionality"
This is a reland of ff292f30d9a4b7a56aea872fe488d342f47202a3

I'm leaving empty .py files in place in order to not break downstream client builds.

Original change's description:
> Remove deprecated barcode scanning functionality
>
> This code is not used anymore, but it's not possible to land this CL
> until issue webrtc:9665 is fixed.
>
> Bug: webrtc:9642,webrtc:9665
> Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
> Reviewed-on: https://webrtc-review.googlesource.com/c/95951
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25289}

TBR=phensman@webrtc.org,phoglund@webrtc.org

Bug: webrtc:9642, webrtc:9665
Change-Id: I248f8656b14c89b0b92e777f4408ee6a6dad41f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107360
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25296}
2018-10-22 21:30:58 +00:00
39feabe35c Enables FrameDecryptor to do an initial key request on frame decryption.
This change enables the FrameDecryptor attached to an RtpVideoReceiver to do
an initial request for a KeyFrame if the first successfully decrypted payload
is not a key frame.

Bug: webrtc:9795
Change-Id: I401ce1f513cb51ce520b60dcaf8b825a68d00c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/107246
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25295}
2018-10-22 21:12:35 +00:00
201596f004 Make packet max buffer size configurable via field trial flag
Bug: webrtc:9851
Change-Id: I2d1960e64d5ff11529317088f838032cecb8ae01
Reviewed-on: https://webrtc-review.googlesource.com/c/107346
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25294}
2018-10-22 16:25:10 +00:00
68b2df7cf1 Make protection_overhead_rate configurable through field trial.
Bug: None
Change-Id: I0a4e03d5f8134809c44d2d0ff88c4b4c8b8a9eaa
Reviewed-on: https://webrtc-review.googlesource.com/c/107341
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25293}
2018-10-22 15:44:23 +00:00
6d2165036c Don't decode frames with an older timestamp than the last decoded timestamp.
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.

Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
2018-10-22 13:11:46 +00:00
38a34198a3 Revert "Remove deprecated barcode scanning functionality"
This reverts commit ff292f30d9a4b7a56aea872fe488d342f47202a3.

Reason for revert: issues with downstream projects

Original change's description:
> Remove deprecated barcode scanning functionality
> 
> This code is not used anymore, but it's not possible to land this CL
> until issue webrtc:9665 is fixed.
> 
> Bug: webrtc:9642,webrtc:9665
> Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
> Reviewed-on: https://webrtc-review.googlesource.com/c/95951
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25289}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,phensman@webrtc.org

Change-Id: I440025777a17d8580526289d4198da1fc3f7d62e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642, webrtc:9665
Reviewed-on: https://webrtc-review.googlesource.com/c/107348
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25291}
2018-10-22 13:00:44 +00:00
67b011d22c Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream
Followup to cl https://webrtc-review.googlesource.com/70880, which
introduced the interface.

Intended to enable tests using MockBitrateAllocator.

Bug: None
Change-Id: I0a784106acf37ff9aca118297233ebd2f2259ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/107342
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25290}
2018-10-22 12:58:33 +00:00
ff292f30d9 Remove deprecated barcode scanning functionality
This code is not used anymore, but it's not possible to land this CL
until issue webrtc:9665 is fixed.

Bug: webrtc:9642,webrtc:9665
Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
Reviewed-on: https://webrtc-review.googlesource.com/c/95951
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25289}
2018-10-22 11:36:40 +00:00
635474e3d5 Compute RTCConnectionState and RTCIceConnectionState.
Compute these states in jseptransportController and store them. Eventually they should be passed on to the peer connection observer and exposed in the blink layer.

Bug: webrtc:9308
Change-Id: Ifdec39c24a607fcb8211c4acf6b9704eaff371b1
Reviewed-on: https://webrtc-review.googlesource.com/c/103506
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25288}
2018-10-22 11:33:17 +00:00
800e121dca Adds support to change transport routes in Scenario tests.
This CL makes it possible to change transport routes while running
a scenario based test.

To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.

Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
2018-10-22 11:14:37 +00:00
8d33c0c104 Adds field trial to do safer reset on route change.
Bug: webrtc:9718
Change-Id: I71143a9616981a24bca7bd5c663a9dae9fc9692e
Reviewed-on: https://webrtc-review.googlesource.com/c/106903
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25286}
2018-10-22 10:46:49 +00:00
c98849cf92 AEC3: changes the signal used for deciding when to update the erle so the reverb render signal is now used
In this CL we change the signal that controls the updates of the ERLE estimator. Until now, the render signal was used which is not optimum for reverberant signals. In this CL, a reverberation has been added to the the render signal and this new signal has been used for controlling when to update the ERLE estimator.

Bug: webrtc:9873
Change-Id: I0ebea3fc208f97aa237af015ba543015d49ed978
Reviewed-on: https://webrtc-review.googlesource.com/c/105660
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25285}
2018-10-22 10:30:12 +00:00
ecdd432a32 Routing unacknowledged data in TransportFeedbackAdapter.
Bug: webrtc:9518
Change-Id: Ie5d016fb5e41645560502af8b8bff324f477229e
Reviewed-on: https://webrtc-review.googlesource.com/c/107302
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25284}
2018-10-22 09:32:11 +00:00
e482ff8f70 Audio codecs API: Remove some weasel words in the docs
These things are no longer brand new, so it makes even less sense
than it once did to warn users that they may change at any time.

Bug: none
Change-Id: I43a6915d9e00fbfef30fdb89869873b129297c8d
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/106980
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25283}
2018-10-22 08:52:15 +00:00
57dd8811c9 Delete dead code in webrtc_libyuv.cc
Delete functions ConvertNV12ToRGB565 and ConvertRGB24ToARGB. Not used,
and not declared in corresponding header file. Left-over from
https://codereview.webrtc.org/2021843002.

Bug: None
Change-Id: I3f95adf52ac8ecdce1089ab79cfac7e1414fe80a
Reviewed-on: https://webrtc-review.googlesource.com/c/106920
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25282}
2018-10-22 08:41:50 +00:00
9acf1c1f54 Reland "Make sure Chromium will pick the correct field_trial/metric impl."
This is a reland of 8dc280d804bc49932f429955b27ff385054ddfa0

Original change's description:
> Make sure Chromium will pick the correct field_trial/metric impl.
> 
> Chromium wants to pick its own field_trial and metrics implementation,
> and WebRTC has an escape hatch to let it remove the definition of these
> symbols. This obviously causes liker errors if Chromium will not pick
> the correct dependency (which cannot be forced by GN check since there
> is no header inclusion).
> 
> Instead of hoping that the build target with the correct implementation
> will be added as a dependency somewhere in the build graph, this CL
> wants to explicitly add the dependency as close as possible to the
> headers, so if they are included and WebRTC is built as part of Chromium
> the correct implementation will be linked into the binary.
> 
> Bug: webrtc:9631
> Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/107040
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25269}

Bug: webrtc:9631
Change-Id: I9b6e6f3f91881e8e3fa8bc6e97f797e8b7e4e4ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107163
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25281}
2018-10-22 07:25:32 +00:00
648d28ad62 Media engine and channel support for per-channel dscp values, specified by RtpParameter
- Similar to network priority
 - Still requires MediaConfig.enable_dscp = true (i.e. googDscp == true to peerconnection)
 - Needs followups 1) Specify value in chrome renderer js idl 2) disable audio bwe when value differs from video  3)remove googDscp guard

Bug: webrtc:5008
Change-Id: Ibdcbb1183f0ca2ae85e3bced6d0aedbccae3ced4
Reviewed-on: https://webrtc-review.googlesource.com/c/93560
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25280}
2018-10-19 21:47:55 +00:00
51cc30c124 Fix a null reference bug in NetworkMonitorAutoDetect.getNetworkState.
Bug: webrtc:9168
Change-Id: Ib3e41db9ff347adfca3b12df6c0fd3293c8ea483
Reviewed-on: https://webrtc-review.googlesource.com/c/107220
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#25279}
2018-10-19 21:18:35 +00:00
cb21ffe899 Add blob-encoding support for RTC event logs
A blob is a string of binary information, whose length may not
necessarily be determined by looking into the string, so that
concatenating all blobs without explicitly including their lengths
as part of their encoding is not a viable option.

Bug: webrtc:8111
Change-Id: I89fdca660e89a6a71eff3ecb7b86416312b81f23
Reviewed-on: https://webrtc-review.googlesource.com/c/104201
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25278}
2018-10-19 17:30:02 +00:00
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

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

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
1cb20deb74 Revert "Make sure Chromium will pick the correct field_trial/metric impl."
This reverts commit 8dc280d804bc49932f429955b27ff385054ddfa0.

Reason for revert: Breaks NDK compile on Chrome bots: https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8932231862774278176/+/steps/compile__with_patch_/0/stdout 
Original change's description:
> Make sure Chromium will pick the correct field_trial/metric impl.
> 
> Chromium wants to pick its own field_trial and metrics implementation,
> and WebRTC has an escape hatch to let it remove the definition of these
> symbols. This obviously causes liker errors if Chromium will not pick
> the correct dependency (which cannot be forced by GN check since there
> is no header inclusion).
> 
> Instead of hoping that the build target with the correct implementation
> will be added as a dependency somewhere in the build graph, this CL
> wants to explicitly add the dependency as close as possible to the
> headers, so if they are included and WebRTC is built as part of Chromium
> the correct implementation will be linked into the binary.
> 
> Bug: webrtc:9631
> Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/107040
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25269}

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

Change-Id: I4c04b60d198aa1b89278083d518bfa93c15d09bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9631
Reviewed-on: https://webrtc-review.googlesource.com/c/107140
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25275}
2018-10-19 14:46:43 +00:00
3c7d599750 Replace _stricmp with absl::EqualsIgnoreCase
All uses check only for equality.

Bug: webrtc:6424
Change-Id: I8755dde02370c89dbc2226bb703664c9e4f88bdb
Reviewed-on: https://webrtc-review.googlesource.com/c/106383
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25274}
2018-10-19 14:17:31 +00:00
53347b7d66 Mute failed tests when no sanitizer defects.
We want sanitizer bots to show failure only for sanitizer defects.
To do so, this CL forces exit code to 0 unconditionally.
Sanitized binaries will turn it to 66 if there is any defect with diagnostic.

Bug: webrtc:9849
Change-Id: I46b683dcae12b76f1be177603af59e3f34bff3a9
Reviewed-on: https://webrtc-review.googlesource.com/c/107060
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25273}
2018-10-19 13:50:35 +00:00
2baa3c49b8 Roll chromium_revision c66210d3ab..b1cb85713b (601019:601125)
Change log: c66210d3ab..b1cb85713b
Full diff: c66210d3ab..b1cb85713b

Changed dependencies
* src/base: 2c31bd007d..703032f2cd
* src/build: 0353cd5458..dc93a673cf
* src/ios: 758462b406..b3f0c9c0fa
* src/testing: 8d815720bc..a02afa425c
* src/third_party: dc690ed2b1..1234d52325
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cedb2de883..a0b50a9a05
* src/tools: 8769652de2..c634df121c
DEPS diff: c66210d3ab..b1cb85713b/DEPS

No update to Clang.

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

Change-Id: I828708f17c0e61de9f177933ecae0a25b67102bc
Reviewed-on: https://webrtc-review.googlesource.com/c/107102
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25272}
2018-10-19 13:32:34 +00:00
0d26c9944c Set renderThreadHandler to null on uncaught exception in EglRenderer.
This should prevent us from posting and deadlocking if EglRenderer
thread crashes.

Bug: b/117400268
Change-Id: I978738249917cb5194917b0b2b12f67bb2a8642e
Reviewed-on: https://webrtc-review.googlesource.com/c/107043
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25271}
2018-10-19 13:16:41 +00:00
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00
8dc280d804 Make sure Chromium will pick the correct field_trial/metric impl.
Chromium wants to pick its own field_trial and metrics implementation,
and WebRTC has an escape hatch to let it remove the definition of these
symbols. This obviously causes liker errors if Chromium will not pick
the correct dependency (which cannot be forced by GN check since there
is no header inclusion).

Instead of hoping that the build target with the correct implementation
will be added as a dependency somewhere in the build graph, this CL
wants to explicitly add the dependency as close as possible to the
headers, so if they are included and WebRTC is built as part of Chromium
the correct implementation will be linked into the binary.

Bug: webrtc:9631
Change-Id: I2d1cfe541726341adfdf81e14e036464aa003d4d
Reviewed-on: https://webrtc-review.googlesource.com/c/107040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25269}
2018-10-19 09:26:25 +00:00
1ddc5b63cc Export symbols needed by the Chromium component build (part 5).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I452117a8385bb08f86c4863bb1079d3774a16a0d
Reviewed-on: https://webrtc-review.googlesource.com/c/107042
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25268}
2018-10-19 09:16:07 +00:00
cf58bf716d Move the SocketStream class to test target
It's used only by the rtc_base/ssladapter_unittest.cc.

Bug: webrtc:9838
Change-Id: Ic89549e1a32003ec0771ca6ba619fecfcf95b493
Reviewed-on: https://webrtc-review.googlesource.com/c/106924
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25267}
2018-10-19 09:15:06 +00:00
bc6a06c058 Adding missing #include on absl/memory/memory.h.
These two files were using absl::make_unique without #including the
header that declares it.

Bug: None
Change-Id: I03019c9a7e06370631680b474d04dd33716b0fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/107041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25266}
2018-10-19 09:14:01 +00:00
82d432980c Delete unused test class StreamSource
Bug: webrtc:6424
Change-Id: Icfb48b997bac71616c67c478f76ab6475a4f92fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106921
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25265}
2018-10-19 08:09:43 +00:00
2461c319ed Roll chromium_revision 343f58e4df..c66210d3ab (600903:601019)
Change log: 343f58e4df..c66210d3ab
Full diff: 343f58e4df..c66210d3ab

Changed dependencies
* src/build: 2d2b19edae..0353cd5458
* src/ios: 69c7749c94..758462b406
* src/testing: 9b5d208818..8d815720bc
* src/third_party: 535cabbec0..dc690ed2b1
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1922eb00bb..cedb2de883
* src/third_party/depot_tools: c1e6594df5..93277a7fc8
* src/tools: ff27d31294..8769652de2
DEPS diff: 343f58e4df..c66210d3ab/DEPS

No update to Clang.

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

Change-Id: I11f12b6c51cf61b6ec96878119505d2f4ca09d69
Reviewed-on: https://webrtc-review.googlesource.com/c/107022
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25264}
2018-10-19 02:39:52 +00:00
97fc11fb86 Fix the 'SetConfiguration(RTCConfiguration::use_media_transport)' setting.
In the past, it would incorrectly set up a state for 'use_media_transport' (i.e. it could say "use_media_transport" is true, but jseptransportcontroller wouldn't know about that).

Also, removes unnecessary field (unused).

Bug: webrtc:9719
Change-Id: I7e5c0ce81b3b70f63c49d661d95b95b5bcbb0c68
Reviewed-on: https://webrtc-review.googlesource.com/c/106960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25263}
2018-10-18 22:29:07 +00:00