Commit Graph

24098 Commits

Author SHA1 Message Date
04255172b6 Remove double declaration of 2 conversion functions.
The declaration in common_types.h is probably a left-over from a
previous cleanup.

Bug: None
Change-Id: I3ee1bad2494ede0022c6aa8fdd106035471d50e2
Reviewed-on: https://webrtc-review.googlesource.com/99220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24666}
2018-09-11 06:00:05 +00:00
49b2c3c4c4 Revert "Decrease complexity of RtpPacketHistory::GetBestFittingPacket."
This reverts commit 54caa4b68a0acb81c7f6ef60ffec45b473a7e1a2.

Reason for revert: Crashes on some perf tests.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc.perf%2FLinux_Trusty%2F7170%2F%2B%2Frecipes%2Fsteps%2Fwebrtc_perf_tests%2F0%2Fstdout



Original change's description:
> Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
> Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.
> 
> Bug: webrtc:9731
> Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
> Reviewed-on: https://webrtc-review.googlesource.com/98882
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24662}

TBR=danilchap@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: Id183cd31a67117e9614d163e4388131fd88de07d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9731
Reviewed-on: https://webrtc-review.googlesource.com/99440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24665}
2018-09-11 05:35:09 +00:00
3a66edf3c3 Use C++11 for range loop in pc/mediasession.cc (and test)
Bug: webrtc:9732
Change-Id: I1fad3313c5ad627f7eca52ea907608d67af6891f
Reviewed-on: https://webrtc-review.googlesource.com/98924
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24664}
2018-09-10 20:27:34 +00:00
f2582da04e Revert "Partial revert of: 'Bump xcode versions for WebRTC bots.'"
This reverts commit 47c48b803202ec0132df06f20c7f16267524ab1a.

Reason for revert: Machines have been upgraded.

Original change's description:
> Partial revert of: 'Bump xcode versions for WebRTC bots.'
> 
> Partial revert of https://webrtc-review.googlesource.com/c/src/+/97060.
> These machines need to be updated to a newer OS version.
> 
> Bug: None
> Change-Id: Ice30ff9125eb366a6d6f93080ae7d0bceba1fe8b
> Reviewed-on: https://webrtc-review.googlesource.com/98400
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24603}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org,oprypin@webrtc.org

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

Bug: None
Change-Id: I7de961d6b6d3b472b0e6bbae545c1902e9a4d0c3
Reviewed-on: https://webrtc-review.googlesource.com/99223
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24663}
2018-09-10 19:25:56 +00:00
54caa4b68a Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.

Bug: webrtc:9731
Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
Reviewed-on: https://webrtc-review.googlesource.com/98882
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24662}
2018-09-10 19:09:29 +00:00
cf64a04acd Bump tryserver mac versions and xcode versions.
Patch by justincohen@, uploaded by phoglund@.

This moves our trybots to 10.13+, which is required for xcode 10,
which is now required by the build scripts.

Bug: None
Change-Id: I77d47bcb6696d290a397b098966ecc4ea1c0aeb9
Reviewed-on: https://webrtc-review.googlesource.com/97301
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24661}
2018-09-10 19:06:12 +00:00
6c092d2993 Refactor initialization of GetRealTimeClock singleton.
This CL also fixes no_exit_time_destructors in system_wrappers.

Bug: webrtc:9693
Change-Id: Ieba752f50949f862244a8348ffc1bed3c2f0150f
Reviewed-on: https://webrtc-review.googlesource.com/99081
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24660}
2018-09-10 18:00:44 +00:00
57606328f6 Adds initial data window field trial to GoogCC.
Bug: webrtc:9718
Change-Id: Ia5a77a09d7ba82b545e9ab12036f717765fdf3b4
Reviewed-on: https://webrtc-review.googlesource.com/97740
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24659}
2018-09-10 16:55:37 +00:00
9be7745509 NetEq tools: Fixing an issue with measuring the simulation time
The NetEqTest class was recently refactored. In the process, the
functionality for measuring the simulation time suffered a bug. This
CL fixes it.

Bug: webrtc:9667
Change-Id: I139e697ede21584ef77ae23cfa8e77f6dac65b51
Reviewed-on: https://webrtc-review.googlesource.com/98982
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24658}
2018-09-10 16:16:22 +00:00
d934244feb Added flags for the adaptive analog AGC in audioproc_f.
Added back the 'agc2 level estimation' flag. Also added a flag for
moving the level measurement before AEC and NS. This is to run offline
experiments with audioproc_f.


Bug: webrtc:7494
Change-Id: I3e3ffceede7166b754130be2b707b620ba527e9f
Reviewed-on: https://webrtc-review.googlesource.com/97442
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24657}
2018-09-10 14:16:46 +00:00
b2d7116733 AEC3: Correction of the suppressor behavior at delay changes
This CL adjusts the behavior of the AEC3 echo suppressor behavior
initially in the call, and when there has been delay changes. The
results is that short echo blips/bursts present in some such cases
no longer occur.

In particular this CL:
-Ensures that the suppressor back-off under stationary render
conditions does not occur until the linear filter has had the
ability to converge.
-Ensures that a previously converged filter behavior detection
is not sticky for stable and linear echo paths, which in turn
prevents echo leakage due to the more liberal echo suppressor
behavior applied on such platforms.
-Removes a bug that caused a random and jittery behavior for
the usage of the linear filter output initially in the calls
and after echo path changes

Bug: webrtc:9737, chromium:882396
Change-Id: Id2b46e366dc58ab8137f19ed59a2034c89ca3087
Reviewed-on: https://webrtc-review.googlesource.com/99063
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24656}
2018-09-10 13:05:14 +00:00
e4f8b38091 Allow different header extensions in 1st packet of a video frame
no behavior changes expected.
Different exension for the 1st packet will be added in a follow-up

Bug: webrtc:9680
Change-Id: I8c853b2710d58df579aeb4b029b42210310423cc
Reviewed-on: https://webrtc-review.googlesource.com/98843
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24655}
2018-09-10 12:53:10 +00:00
abd4273e43 Cleanup test::FrameGeneratorCapturer::InsertFrameTask
Fix race between FrameGeneratorCapturer destructor and inserting a frame.
(aka do not use this after reposting self to a TaskQueue)

Separate periodic case from one-time case.
When Generator can't keep up with fps, drop frames instead of trying to catch up.
Use absl::make_unique and absl::WrapUnique

Bug: None
Change-Id: I9d5d1fcacf174e28d83310099b79e26ece9b7b37
Reviewed-on: https://webrtc-review.googlesource.com/98844
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24654}
2018-09-10 12:35:53 +00:00
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