Commit Graph

23885 Commits

Author SHA1 Message Date
7e5203f151 Revert "Reenable simulcast video full stack test on MAC"
This reverts commit 2c446f2723451011f8c3ec5d240337e9e309a577.

Reason for revert: Still hangs, e.g. here: https://ci.chromium.org/buildbot/client.webrtc.perf/Mac%2010.11/7204

Original change's description:
> Reenable simulcast video full stack test on MAC
>
> Reenabling test because a possible hanging cause was fixed here:
> https://webrtc-review.googlesource.com/c/src/+/96980
>
> Bug: webrtc:9220
> Change-Id: I74243eeebe5646c54373fba04bff27d456df7771
> Reviewed-on: https://webrtc-review.googlesource.com/98500
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24624}

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

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

Bug: webrtc:9220
Change-Id: I172ecec57233131267e38b0221564e6f3f88941f
Reviewed-on: https://webrtc-review.googlesource.com/99180
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24653}
2018-09-10 12:33:13 +00:00
067818fe85 Move RtcpTransceiver deletion of copy and assign methods to public section
Bug: chromium:881453
Change-Id: Iff5c522b983af018c1308649887a1121519c73ea
Reviewed-on: https://webrtc-review.googlesource.com/98981
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24652}
2018-09-10 12:23:19 +00:00
52b4961ae1 Disallow assign by deleting correct assign signature
Bug: chromium:881453
Change-Id: I80e74d0ed37d98b3472a31a42c3468f1bdbbb950
Reviewed-on: https://webrtc-review.googlesource.com/99061
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24651}
2018-09-10 12:22:14 +00:00
8cec4fb6c2 Use default RTCConfiguration on iOS
With "aggressive" preset the default bundlePolicy is set to "maxBundle" when it shoud be "balanced" according to spec.

Bug: webrtc:9458
Change-Id: Ifbdd76be3a6d9968574cba857f178d5f859dcb87
Reviewed-on: https://webrtc-review.googlesource.com/88567
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24650}
2018-09-10 12:16:53 +00:00
7f978f18b3 Revert "Enable VP9 KSVC perf tests."
This reverts commit 1fdcfa755e963e2710fcbf6b1525b6ed50e67428.

Reason for revert: Still crashes (see webrtc:9506).

Original change's description:
> Enable VP9 KSVC perf tests.
> 
> The tests crashed and were disabled temporarily. The crash was probably
> caused by chromium:879307 which was fixed recently.
> 
> Bug: webrtc:9506
> Change-Id: I08872c0370c9cf5dc4769daf68b7c61135a55c9e
> Reviewed-on: https://webrtc-review.googlesource.com/99080
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24638}

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

Change-Id: I4761c702330809f0e39e6a88870892320dc47280
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9506
Reviewed-on: https://webrtc-review.googlesource.com/99160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24649}
2018-09-10 12:02:14 +00:00
a94ad2920c Bump iOS bots to iOS 11.
This should also solve the trigger problem on the iOS perf bot.

Remove device_type which appears to be ignored anyway. For instance,
device_type said iphone 6s but we got iPhone 8 when I actually looked
in the swarming dimension.

Bug: webrtc:7156
Change-Id: I1aa22e7f217deebf9eeee18363622e37ecc2a40e
Reviewed-on: https://webrtc-review.googlesource.com/99060
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24648}
2018-09-10 12:00:26 +00:00
623472219f Store RuntimeSetting in Aec Dumps.
Also read and apply settings when parsing and replaying dumps.

The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
  AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
  audioproc_f.

Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 11:40:28 +00:00
042661b404 Revert "Frame rate controller per spatial layer."
This reverts commit ae9e188e67a489db597224e3cfcfdee04edf0cba.

Reason for revert: Verify if this causes chromium:882358.

Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}

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

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

Bug: webrtc:9682, chromium:882358
Change-Id: Idc4051eef72104823038ed9139bb9c75018f7d86
Reviewed-on: https://webrtc-review.googlesource.com/99082
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24646}
2018-09-10 11:24:33 +00:00
fb2a66a58a libvpx vp8 encoder: get frame drop setting from temporal layer
Today, the internal frame dropper in libvpx vp8 encoder is enabled or
disabled based on video or screen content. This is then expected to
match up with screenshare vs default temporal layers implementation.

This cl makes libvpx query the temporal layers implementation as well,
breaking this implicit dependency and allows frames to be dropped if
default temporal layers is used with screen content.

Bug: webrtc:9734
Change-Id: If2523a211f4929f16e65a02fa7a6b4edf7328571
Reviewed-on: https://webrtc-review.googlesource.com/99062
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24645}
2018-09-10 11:10:11 +00:00
0417eadbf2 Removed unused member |last_unwrap_| from RtpFrameReferenceFinder.
Bug: none
Change-Id: Ideb876d89dbab7a9f4c8c46d95217f00e07b62d6
Reviewed-on: https://webrtc-review.googlesource.com/98862
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24644}
2018-09-10 10:03:41 +00:00
4384f53285 Add more useful information to NetEqState and implement action_times_ms
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.

Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
2018-09-10 09:10:53 +00:00
7461eff1bd For simulcast screenshare, make 2 tl default for high stream.
Bug: webrtc:9734
Change-Id: I00400782686296b191f0f7a10a65f99253bea929
Reviewed-on: https://webrtc-review.googlesource.com/99101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24642}
2018-09-10 09:07:59 +00:00
76dac9ac2f Fix no_global_constructors in modules/video_capture.
Bug: webrtc:9693
Change-Id: Ia917ab824f18991cfdcffa04ea9c063c6a224532
Reviewed-on: https://webrtc-review.googlesource.com/98640
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24641}
2018-09-10 09:03:39 +00:00
211856b956 Make HasAttribute handle partial matching of attribute names.
Improve HasAttribute to handle the case where the beginning of an
attribute name is also an attribute name in it self. Two attributes
that have this relation are extmap-allow-mixed and extmap.

Bug: webrtc:9712
Change-Id: Iee660cc6e3dc7f2e7c56664a4f0ffb298eca9208
Reviewed-on: https://webrtc-review.googlesource.com/97422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24640}
2018-09-10 08:43:52 +00:00
2903888cde Verify posting task and reply just before task queue destruction
Bug: webrtc:9728
Change-Id: I516311a507b4e9f49c45fda5185e96d4248ed455
Reviewed-on: https://webrtc-review.googlesource.com/98520
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24639}
2018-09-10 08:40:35 +00:00
1fdcfa755e Enable VP9 KSVC perf tests.
The tests crashed and were disabled temporarily. The crash was probably
caused by chromium:879307 which was fixed recently.

Bug: webrtc:9506
Change-Id: I08872c0370c9cf5dc4769daf68b7c61135a55c9e
Reviewed-on: https://webrtc-review.googlesource.com/99080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24638}
2018-09-10 08:30:14 +00:00
b33290d082 Return null from PCFactory#createPeerConnection on failure.
Currently, invalid PeerConnection object is returned. With this change,
null is returned instead. This can be more easily handled in the
application layer.

Bug: webrtc:9440
Change-Id: I44dfee81a681f033b8d336c999d43ff1c69fb015
Reviewed-on: https://webrtc-review.googlesource.com/98480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24637}
2018-09-10 07:13:40 +00:00
4daf66e71e Use C++11 style for loop in webrtcsdp.cc
Bug: webrtc:9732
Change-Id: I76eda7cca0aab6989bea819bfff4e06034c399f7
Reviewed-on: https://webrtc-review.googlesource.com/98922
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24636}
2018-09-07 23:17:04 +00:00
0b445c6271 Cleanup RtpPacketizerVp9
Merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If230a4ea901b5cdd6a376f8dd2db48e94d6dca36
Reviewed-on: https://webrtc-review.googlesource.com/98866
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24635}
2018-09-07 22:01:46 +00:00
5af3051f84 Fix no_exit_time_destructors in ortc.
Non trivially destructible objects with static storage are disallowed
by the style guide.

This CL just removes 'static' since these objects are constructed once
or twice in the entire application.

Bug: webrtc:9693
Change-Id: I7509e2c088dd5ec0ac13f08053ecb76cf8259d90
Reviewed-on: https://webrtc-review.googlesource.com/98840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24634}
2018-09-07 20:47:15 +00:00
d7b79af9df Add "tones remaining" argument to DTMF ontonechange callback
Bug: webrtc:9725
Change-Id: I2ad3e57d7357a9bd7cfbfa675df36ec66ff7c851
Reviewed-on: https://webrtc-review.googlesource.com/98361
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24633}
2018-09-07 17:29:37 +00:00
792df6b4b9 Make RtcpTransceiver destructor non-blocking
At cost of removing assumption callbacks can't be used after destructor.

Bug: webrtc:8239
Change-Id: Id79f7553528cf6c102d3ee0bf7aa2de5b0437d2a
Reviewed-on: https://webrtc-review.googlesource.com/98860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24632}
2018-09-07 15:34:08 +00:00
d7027dc081 Revert "Fix no_global_constructors in audio_processing/agc2/rnn_vad."
This reverts commit 5e2e66d8a0fd5e1bf9b3efc54a94cba3e7088b00.

Reason for revert: Change implementation.

Original change's description:
> Fix no_global_constructors in audio_processing/agc2/rnn_vad.
> 
> Bug: webrtc:9693
> Change-Id: Ica997d5cbe28288720325a51058a40a37c612665
> Reviewed-on: https://webrtc-review.googlesource.com/98583
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24617}

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

Change-Id: I9e30f6ec08baa22a8d6c15546341000738c095b6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9693
Reviewed-on: https://webrtc-review.googlesource.com/98842
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24631}
2018-09-07 13:34:39 +00:00
d8c50780ea Reland "Consolidate loggability checks and replace streams."
Currently we check if a message should be printed at the call site using LogMessage::Loggable, in the LogMessage itself using LogMessage::IsNoop and in LogMessage::OutputToDebug using log_to_stderr_.

This change unifies the first two of these into a early return in Log().

Bug: webrtc:8982
Change-Id: I462b1cf63c44fec46e5c59b147b2b99605aaae0c
Reviewed-on: https://webrtc-review.googlesource.com/98820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24630}
2018-09-07 13:27:29 +00:00
d52a1a6971 Reland "Remove RTPVideoHeader::vp8() accessors."
This reverts commit 1811c04f22a26da3ed2832373a5c92a9786420c3.

Reason for revert: Downstream projects fixed.

Original change's description:
> Revert "Remove RTPVideoHeader::vp8() accessors."
> 
> This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.
> 
> Reason for revert: Break downstream projects.
> 
> Original change's description:
> > Remove RTPVideoHeader::vp8() accessors.
> > 
> > Bug: none
> > Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> > Reviewed-on: https://webrtc-review.googlesource.com/93321
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24626}
> 
> TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com
> 
> Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: none
> Reviewed-on: https://webrtc-review.googlesource.com/98864
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24628}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I9246f36e638108ae4fc46c1ae4559c8205d50fc1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24629}
2018-09-07 13:04:07 +00:00
1811c04f22 Revert "Remove RTPVideoHeader::vp8() accessors."
This reverts commit af7afc66427b0e9109e7d492f2805d63d239b914.

Reason for revert: Break downstream projects.

Original change's description:
> Remove RTPVideoHeader::vp8() accessors.
> 
> Bug: none
> Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
> Reviewed-on: https://webrtc-review.googlesource.com/93321
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24626}

TBR=danilchap@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,holmer@google.com

Change-Id: I3f7f19c0ea810c0fd988c59e6556bbea9b756b33
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/98864
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24628}
2018-09-07 12:36:17 +00:00
5470f4030f Renamed GetCodecHeader to GetRtpVideoHeader in RtpFrameObject.
Bug: none
Change-Id: I158a19dc85ef12dc86f603ff0f6618b89cb1c242
Reviewed-on: https://webrtc-review.googlesource.com/98863
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24627}
2018-09-07 12:35:12 +00:00
af7afc6642 Remove RTPVideoHeader::vp8() accessors.
Bug: none
Change-Id: Ia7d65148fb36a8f26647bee8a876ce7217ff8a68
Reviewed-on: https://webrtc-review.googlesource.com/93321
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24626}
2018-09-07 12:01:19 +00:00
5e007b77f1 Use function-local static variable for MessageQueueManager singleton.
Rely on C++11 thread-safe initialization on first call to
MessageQueueManager::Instance(), in the same way as for
ThreadManager::Instance().

Bug: None
Change-Id: I26244f90c5d7f94a2454688297f55bf96617e78c
Reviewed-on: https://webrtc-review.googlesource.com/97721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24625}
2018-09-07 11:58:17 +00:00
2c446f2723 Reenable simulcast video full stack test on MAC
Reenabling test because a possible hanging cause was fixed here:
https://webrtc-review.googlesource.com/c/src/+/96980

Bug: webrtc:9220
Change-Id: I74243eeebe5646c54373fba04bff27d456df7771
Reviewed-on: https://webrtc-review.googlesource.com/98500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24624}
2018-09-07 10:48:30 +00:00
2e4419e05b Add option to only request a frame interval change via OnOutputFormatRequest.
OnOutputFormatRequest(const absl::optional<VideoFormat>& format)
changed to
OnOutputFormatRequest(
      const absl::optional<std::pair<int, int>>& target_aspect_ratio,
      const absl::optional<int>& max_pixel_count,
      const absl::optional<int>& max_fps)

Decouples:
- Resolution and fps requests.
- Resolution requests from aspect ratio requests.

Bug: webrtc:9597
Change-Id: I6f44c91283cf5474c6531e55773d2257e2341063
Reviewed-on: https://webrtc-review.googlesource.com/95423
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24623}
2018-09-07 10:36:27 +00:00
2f864fb4ab Handle empty GOF.
Assume that stream has single temporal layer if number of frames in GOF
is set to zero (valid case).

Bug: chromium:879584
Change-Id: I7ced082190e40c1bf4cc1468babfd98b0a61f0dd
Reviewed-on: https://webrtc-review.googlesource.com/98800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24622}
2018-09-07 10:19:06 +00:00
3445b6bc51 Roll chromium_revision d03e9be2ec..de6484608c (589368:589477)
Change log: d03e9be2ec..de6484608c
Full diff: d03e9be2ec..de6484608c

Changed dependencies:
* src/base: 9ae2c8c4c9..2e3b697294
* src/ios: 9f1665f0ba..69ec23fc6a
* src/testing: dcc549489a..ea02f4bb3f
* src/third_party: b699212fd8..b45a62004d
* src/third_party/depot_tools: b56a43a906..515e7fe037
* src/tools: 96ea619391..5999232ae4
DEPS diff: d03e9be2ec..de6484608c/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ic77d940ac40f53e56bbb83dfb3fa73b357f37855
Reviewed-on: https://webrtc-review.googlesource.com/98769
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24621}
2018-09-07 10:07:21 +00:00
fe3240aeae Reland "Delete class EventTimerWrapper."
This is a reland of a421775a6d4f78f7aa9c3ea020a8834e049efbcc

Original change's description:
> Delete class EventTimerWrapper.
>
> Only user, iSACTest, refactored to use a sleep instead.
>
> Bug: webrtc:3380
> Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
> Reviewed-on: https://webrtc-review.googlesource.com/96802
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24541}

Tbr: henrik.lundin@webrtc.org
Bug: webrtc:3380
Change-Id: I541473b9c3ce2020f76d420598a7b10766f1d2a9
Reviewed-on: https://webrtc-review.googlesource.com/98481
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24620}
2018-09-07 09:54:55 +00:00
6ad6e1f04c Removed old and unused WebRTC-NewVideoJitterBuffer field trial from VideoReceiveStreamTest.
Bug: none
Change-Id: Ide15295feb8ebba71a11ac083f8ca84902c4d24c
Reviewed-on: https://webrtc-review.googlesource.com/98560
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24619}
2018-09-07 09:49:59 +00:00
fa5ec8d20d Use signed integers for limiting packet size in video packetizers
Using signed integers allows to centralize checking of edge cases
in RtpPacketizer::SplitAboutEqually and
reduce chance of hitting issues with size_t underflow

Bug: webrtc:9680
Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08
Reviewed-on: https://webrtc-review.googlesource.com/98602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24618}
2018-09-07 09:24:18 +00:00
5e2e66d8a0 Fix no_global_constructors in audio_processing/agc2/rnn_vad.
Bug: webrtc:9693
Change-Id: Ica997d5cbe28288720325a51058a40a37c612665
Reviewed-on: https://webrtc-review.googlesource.com/98583
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24617}
2018-09-07 08:08:45 +00:00
7cd2a95148 Roll chromium_revision 450316fffc..d03e9be2ec (589248:589368)
Change log: 450316fffc..d03e9be2ec
Full diff: 450316fffc..d03e9be2ec

Changed dependencies:
* src/base: 3e92bda546..9ae2c8c4c9
* src/build: ebaffbc692..6533d0538d
* src/ios: 589b5014c9..9f1665f0ba
* src/testing: 2030744249..dcc549489a
* src/third_party: 6adbdba00a..b699212fd8
* src/third_party/depot_tools: 0f5a0b4409..b56a43a906
* src/third_party/libyuv: d694f0a82b..9a07219dc8
* src/tools: 901e527957..96ea619391
DEPS diff: 450316fffc..d03e9be2ec/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I303c7fc605a7b5e72ecc70021e057c60a2532dda
Reviewed-on: https://webrtc-review.googlesource.com/98706
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24616}
2018-09-07 00:09:53 +00:00
ed425d12c4 Multiplex codec cleanups
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.

Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
2018-09-06 23:29:15 +00:00
6a4fd19bbd AEC3: Parametrize the delay estimator to leverage strong echo paths
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.

To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.

Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
2018-09-06 23:01:58 +00:00
88e3e3f570 Updated FrameEncryptorInterface and FrameDecryptorInterface with status code.
This change allows an implementer to provide a custom error code to be returned
on failure with 0 being reserved for success. It also modifies the output size
APIs to be more specific and requests the number of bytes written to be returned
so we can determine how many bytes we need to send.

Bug: webrtc:9681
Change-Id: I13d34861bf851527fcbb350d0cfb480c0f95a6b3
Reviewed-on: https://webrtc-review.googlesource.com/98720
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24613}
2018-09-06 21:08:59 +00:00
f60bd4bb00 Interface for media transport
This is experimental interface for media transport.

The goal is to refactor WebRTC codebase to send/receive frames via media transport interface. It will allow us to have different media transport implementations in the future, including QUIC-based media transport.

Bug: webrtc:9719
Change-Id: I64e0b69d18c212e1ed0a08c6904578c3dfbe3af7
Reviewed-on: https://webrtc-review.googlesource.com/95960
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24612}
2018-09-06 20:15:22 +00:00
3c8d3e4fb2 Roll chromium_revision fcd25fc0ff..450316fffc (589073:589248)
Change log: fcd25fc0ff..450316fffc
Full diff: fcd25fc0ff..450316fffc

Changed dependencies:
* src/base: ecfe9ef251..3e92bda546
* src/ios: bc1e024915..589b5014c9
* src/testing: ce1bea4616..2030744249
* src/third_party: 86f33d4bca..6adbdba00a
* src/third_party/depot_tools: 01f0c698fb..0f5a0b4409
* src/tools: 7c7257845b..901e527957
DEPS diff: fcd25fc0ff..450316fffc/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I386fd72ad65a7f4660b19e139c7f602dd2be6b52
Reviewed-on: https://webrtc-review.googlesource.com/98702
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24611}
2018-09-06 20:13:48 +00:00
ea8b6f95c7 Adds the Java interface points for FrameEncryptor/FrameDecryptor.
This changes adds the API surface for injecting the FrameEncryptor and FrameDecryptor from Java.
This assumes that the API User will be able to provide native implementations of both the Encryptor
and Decryptor. Optional Java implementations may come later but due to the significant performance
issues around copying every frame across the JNI boundary it doesn't seem like a good idea to support
a non native backed implementation for now.

Bug: webrtc:9681
Change-Id: Ib4471e69fdf0a99705f824de652c621637b92326
Reviewed-on: https://webrtc-review.googlesource.com/96865
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24610}
2018-09-06 19:41:21 +00:00
c003a1f067 Fix no_global_constructors in congestion_controller/pcc.
Bug: webrtc:9693
Change-Id: I7d654656b4f350c120d25dcc0f66541f47ccc919
Reviewed-on: https://webrtc-review.googlesource.com/98582
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24609}
2018-09-06 16:18:19 +00:00
59dd972e37 Fix no_global_constructors in congestion_controller/bbr.
Bug: webrtc:9693
Change-Id: I47eb1b27adb0fd40e7955e477fa31cdc462891cd
Reviewed-on: https://webrtc-review.googlesource.com/98581
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24608}
2018-09-06 16:16:09 +00:00
50cfab7d84 Fix no_global_constructors in congestion_controller/goog_cc.
Bug: webrtc:9693
Change-Id: Ic341f3bed2c7c8c4f62363a7344f8524ff561155
Reviewed-on: https://webrtc-review.googlesource.com/98580
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24607}
2018-09-06 16:01:49 +00:00
95c56eebe0 Fix paths for macOS framework’s copy actions.
Using the mac framework bundle's path should be more correct than the
hardcoded version used previously.

Bug: webrtc:9627
Change-Id: I706c7961f21ba921503ed68b6e79ea0838e664d5
Reviewed-on: https://webrtc-review.googlesource.com/98620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24606}
2018-09-06 14:13:11 +00:00
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
96ede16a4e Enable -Wexit-time-destructors and -Wglobal-constructors.
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.

It also adds the possibility to suppress these warnings because
they trigger in a few places.

The long term goal is to avoid regressions on this and remove all the
suppressions.

Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
2018-09-06 12:43:20 +00:00