3633 Commits

Author SHA1 Message Date
Erik Språng
54d5d2c75b Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.

The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.

Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
2019-08-21 09:45:21 +00:00
Bryan Ferguson
e8ef87bdad Include menus & dialogs in frames captured by WindowCapturerWin
This change adds logic to WindowCapturerWin to capture overlapping
owned/pop-up windows (e.g. menus, dialogs, tooltips). This makes window
capture behavior more consistent regardless of whether
CroppingWindowCapturerWin is used & its conditions for using crop-from-
screen capture are met (in ShouldUseScreenCapturer). (I.e. regardless
of OS version, window shape / translucency, occlusion by another
potentially top-most window, or whether the capturing app has opted in
to using the cropping capturer).

Owned/pop-up windows associated with the selected window are enumerated
then captured individually, with their contents composited into the
final frame.

This change also:
- Crops out the top window border (which exposed a bit of the background
  when using the cropping capturer, and resulted in an inconsistent
  appearance compared to the side & bottom borders being cropped out).

Bug: chromium:980864
Change-Id: I81c504848a0c0e6bf122aeff437b400e44944718
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148302
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28922}
2019-08-21 07:55:07 +00:00
Mirko Bonadei
2dac4e4e35 Remove rtc_use_lto GN arg.
This CL is a no-op since rtc_use_lto is always false and in general
such change should probably be implemented in
//build/config/compiler/BUILD.gn.

Bug: chromium:408997
Change-Id: Id37d3181e66e699f8cd535aee1af7609352a7259
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149833
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28919}
2019-08-20 14:00:49 +00:00
Niels Möller
5ceb4ac5ed Delete some unused AudioCodingModule methods
Methods deleted:

  ReceiveFrequency, PlayoutFrequency, ReceiveCodec,
  SetMinimumPlayoutDelay, SetMaximumPlayoutDelay,
  SetBaseMinimumPlayoutDelayMs, GetBaseMinimumPlayoutDelayMs,
  PlayoutTimestamp, FilteredCurrentDelayMs, TargetDelayMs.

Became unused with cl
https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: Ie50e8e86a622661c3daa9db83a2e66489dcd2d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148071
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28918}
2019-08-20 13:42:36 +00:00
Niels Moller
e21f3f574b Revert "Delete mac_utils.h and mac_utils.cc"
This reverts commit ada8e17125d2124f5bcdc1558182ce95d6311d93.

Reason for revert: Breaks chromium, due to undeclared dependency on SystemConfiguration.framework

Original change's description:
> Delete mac_utils.h and mac_utils.cc
> 
> They defined two functions: ToUtf16 and ToUtf8. The former was unused,
> and the latter is moved to
> modules/desktop_capture/mac/window_list_utils.cc, the only user.
> 
> Tbr: sergeyu@chromium.org
> Bug: None
> Change-Id: Ib8a513da42e43ba8d41a2de4c1645b3f48448dc9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148531
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sergey Ulanov <sergeyu@google.com>
> Cr-Commit-Position: refs/heads/master@{#28913}

TBR=zijiehe@chromium.org,nisse@webrtc.org,kthelgason@webrtc.org,sergeyu@google.com,sergeyu@chromium.org

Change-Id: I9d6a2f63b3acde0eefab92d034529b800d6adcab
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149811
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28915}
2019-08-20 09:58:37 +00:00
Niels Möller
ada8e17125 Delete mac_utils.h and mac_utils.cc
They defined two functions: ToUtf16 and ToUtf8. The former was unused,
and the latter is moved to
modules/desktop_capture/mac/window_list_utils.cc, the only user.

Tbr: sergeyu@chromium.org
Bug: None
Change-Id: Ib8a513da42e43ba8d41a2de4c1645b3f48448dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148531
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sergey Ulanov <sergeyu@google.com>
Cr-Commit-Position: refs/heads/master@{#28913}
2019-08-20 08:52:28 +00:00
Per Åhgren
928146f546 Removing all external access to the integer sample data in AudioBuffer
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.

The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.

Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
2019-08-20 08:36:47 +00:00
Konrad Hofbauer
fdf38802a6 Make "WebRTC-BweAllocProbingOnlyInAlr/Enabled/" default and remove key.
Bug: chromium:951299
Change-Id: Idf612040e21f2962cc63d7de3dcb237bbf868034
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148985
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28902}
2019-08-19 15:39:25 +00:00
Erik Språng
185243b335 Remove most of PacedSenderUnittest
These tests are now run as part of PacingControllerUnittest instead.

Bug: webrtc:10809
Change-Id: If59e622e8a66565be678106d9341aa6eee78c299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149803
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28900}
2019-08-19 13:32:59 +00:00
Per Åhgren
62c174c5a1 Reland of Correct conversion between float and fixed formats
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.

In contrast to the initial CL, the DCHECKS on the incoming sample
range was changed to limiting.

Bug: webrtc:6594
Change-Id: I8218dfd5c45388ad5aac61be453d2f28725a2475
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#28867}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149169
Cr-Commit-Position: refs/heads/master@{#28897}
2019-08-19 12:15:07 +00:00
Erik Språng
93f518917f Remove some usage of RtpRtcp::SetSSRC()
Bug: webrtc:10774
Change-Id: Ib8fa84f5d70ceb7e715405eae2901bcd7bdfebfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28895}
2019-08-19 11:11:41 +00:00
Gustaf Ullberg
cd277b84da AEC3: Fix computation of audio buffer delay
This change fixes a bug where the initial delay could be set incorrectly.

Bug: webrtc:10896
Change-Id: I66b2234b69c46639488f4561e973384001230861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28894}
2019-08-19 11:05:21 +00:00
Danil Chapovalov
83773b555c Delete deprecated RtpRtcp::CreateRtpRtcp factory
Bug: None
Change-Id: I2ace74c380b89d300a6d0e7cca4766147f33cb1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149821
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28890}
2019-08-19 10:07:10 +00:00
Erik Språng
d05edecf4c Extract most of PacedSender into PacedSendingController.
The Pacer now just handles interaction with Module/ProcessThread and
forwarding packets to PacketRouter.
All other logic is moved to PacedSendingController, including tests.
PacedSender unittest are now just some basic sanity tests.

Bug: webrtc:10809
Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28886}
2019-08-19 09:07:28 +00:00
Ying Wang
0e1a558fb3 Allowing 40ms audio frame length.
Currently 20ms, 60ms and 120ms frame length are supported. The motivation is to better adapt audio bit rate to network conditions with more frame length choices.

This is continuation of https://webrtc-review.googlesource.com/c/src/+/146206, since crodbro is out of office, I created this commit for continuing the code review.

Bug: webrtc:10820
Change-Id: I0e35e91b524f63686bfdf767b7a95c51aeb24716
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146780
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28882}
2019-08-16 20:24:18 +00:00
Åsa Persson
f5e5d250bc BalancedDegradationSettings: add option to configure a min framerate diff.
If a framerate reduction (input fps - restricted fps) is less than the
configured diff, shorten interval to next qp check.

Bug: none
Change-Id: Ia0b9e0638e5ba75cdc20a1bb45bfcb7d858c5f89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149040
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28880}
2019-08-16 16:13:46 +00:00
Alex Narest
bbeb10925e Reporting audio device underrun counter
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
2019-08-16 11:49:55 +00:00
Minyue Li
9b29d69650 Make ANA frame length controller more robust to encoder frame lengths.
Bug: webrtc:10820
Change-Id: Ic3a30976d0181de9cdd35e44d4c5439cadad4812
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28873}
2019-08-16 10:55:39 +00:00
Henrik Andreassson
533c225c93 Revert "Correct conversion between float and fixed formats"
This reverts commit 67e43c8b95057a889ba9946e47d50a265e1e9ac9.

Reason for revert: speculative revert since we see failing bots on Android after this change

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/4124

Original change's description:
> Correct conversion between float and fixed formats
> 
> This CL changes the way that values are converted
> between fixed and floating point to
> -Avoid the former asymmetric conversion causing
> nonlinear distortions.
> -Reduce the complexity.
> 
> Bug: webrtc:6594
> Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28867}

TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org,peah@webrtc.org

Change-Id: Id828a09de7075e48556fe2d0beba7a0c6ec227f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149165
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28872}
2019-08-16 10:40:11 +00:00
Per Åhgren
67e43c8b95 Correct conversion between float and fixed formats
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.

Bug: webrtc:6594
Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28867}
2019-08-15 13:51:39 +00:00
Per Åhgren
a1351271e6 Remove all AudioBuffer code that is not related to storing audio data
This CL moves/removes all code from the AudioBuffer that:
-Is not directly handling audio data (e.g., keytaps, VAD descisions).
-Is caching aggregated versions of the rest of the audio data.
-Is not used (or only used in testing)

Bug: webrtc:10882
Change-Id: I737deb3f692748eff30f46ad806b2c6f6292802c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149072
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28866}
2019-08-15 13:49:29 +00:00
Sebastian Jansson
3aa0d76cb0 Use struct parser for AlrDetector config.
Bug: webrtc:9883
Change-Id: Ib58fa5ba87607a268f4960898625b1a5adcab69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148596
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28862}
2019-08-14 18:23:05 +00:00
Minyue Li
c759f832e9 Avoid copying of vectors in RtpPacketInfos.
Bug: chromium:982260
Change-Id: Ia4dab497b662e825f80c16530cdf615b62f0a5c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148523
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28859}
2019-08-14 15:46:02 +00:00
Yves Gerey
704c8c4446 Re-enable AudioDeviceTest in combination with sanitizers.
Reactivate all tests which aren't flaky anymore.

Bug: webrtc:9751, webrtc:10867
Change-Id: I1d76e0f3e6cc82e78fc46214202f40a9666d41fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149060
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28853}
2019-08-14 08:45:18 +00:00
Niels Möller
78c56cba00 Delete deprecated version of ReceiveStatistics::Create
Bug: webrtc:10679
Change-Id: I885f38a80c0fe10f1596f33fa95e40a91b23001c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148445
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28852}
2019-08-14 08:40:09 +00:00
Jiawei Ou
608e6ba394 Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.

Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28847}
2019-08-14 00:40:19 +00:00
Niels Möller
d78196576d Delete StreamDataCountersCallback from ReceiveStatistics
Bug: webrtc:10679
Change-Id: Ife6a4f598c5b70478244b15fc884f6a424d1505b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148521
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28841}
2019-08-13 14:47:08 +00:00
Niels Möller
01525f9e03 Delete method StreamStatistician::GetDataCounters
Usage replaced with GetReceiveStreamDataCounters.

Bug: None
Change-Id: Ic5f62ff8a8d33b9eec21657512ba6a0a44635e6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28840}
2019-08-13 13:46:45 +00:00
Erik Språng
82d75a6214 Use unit types in RoundRobingPacketQueue and PacedSender
This CL replaces various int types with DataRata, DataSize, Timestamp
and TimeDelta classes.

This is part of larger refactoring work where most of PacedSender will
be broken out into a class handling the logic and another responsible
for thread handling. Splitting that up for easier reviewing.

Bug: webrtc:10809
Change-Id: If57a238e5090c47bf3a99c2042783ae584b425f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148591
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28835}
2019-08-12 17:10:21 +00:00
Yves Gerey
110a4de4e2 Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691)
!! **Manual change** Less strict audio codec tests to accommodate opus fix [1].
!! This is meant to be a temporary mitigation.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/1746617

Change log: 8f0166a59b..f0fd984a31
Full diff: 8f0166a59b..f0fd984a31

Changed dependencies
* src/base: 17d8ac209c..f6cc884505
* src/build: d6837de8f1..956965a6ea
* src/ios: 76e0b0bc60..6780db9c3e
* src/testing: 5d328647a1..48823ed18a
* src/third_party: d70201c684..82063e79f0
* src/third_party/depot_tools: 1b4c7e9f38..6d98232fde
* src/tools: b8953a5bf5..2aa12eadc5
DEPS diff: 8f0166a59b..f0fd984a31/DEPS

No update to Clang.

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

Change-Id: I9869cc3f493bc82361e4f93ad846b32390edb340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148700
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28833}
2019-08-12 15:53:01 +00:00
Niels Möller
58b496b4d8 Let StreamStatistician::GetReceiveStreamDataCounters return counters by value
Tbr: ossu@webrtc.org # Trivial update of audio/ call site
Bug: None
Change-Id: I3763e83f6c0e18be1b696dd7b2ba5841045c9159
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148820
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28830}
2019-08-12 12:36:00 +00:00
Yves Gerey
412282acf9 [tsan] Guard audio_device_pulse_linux members from concurrent access.
This CL also fixes data races caused by tests themselves.

TBR= henrika@webrtc.org

Bug: webrtc:9751
Change-Id: Ie7c785b27142fd465f5b4dc9fb0628bd7274f1d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146600
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28829}
2019-08-12 11:55:52 +00:00
Erik Språng
1691e88584 Remove unused fallback method in PacedSender
Bug: webrtc:10809
Change-Id: I30279082c9fa616a686259eb1efc0ebcc1819f61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148592
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28828}
2019-08-12 10:33:48 +00:00
Niels Möller
dc5ed5c023 Delete NACK-related methods from AudioCodingModule
Unused since cl https://webrtc-review.googlesource.com/c/src/+/111504

Bug: None
Change-Id: I210f9c286961a2aec73c7e5c4cf8d04160f5a190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148076
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28827}
2019-08-12 09:41:10 +00:00
Sonia-Florina Horchidan
b75d14c802 audioproc_f: input AEC dump as string, output audio to vector
This CL adds the following options:

pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector

Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
2019-08-12 09:17:36 +00:00
Jakob Ivarsson
81df62b456 Add field trial to introduce extra delay after target level calculation.
Bug: webrtc:10817
Change-Id: Id9eced821df2859b2cb7174062b6f5e29e145f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145902
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28825}
2019-08-12 08:55:23 +00:00
Bryan Ferguson
1544915bb4 Avoid capturing extraneous windows in CroppingWindowCapturerWin
This change reduces cases where capturing a window with the cropping
capturer captures unrelated windows from the same process. For instance:
- Capturing an Explorer window could include portions of taskbar UI,
  e.g. when an auto-hide taskbar or window preview thumbnails are shown
  overtop.
- Capturing a window from a process with multiple windows could include
  menus/tooltips from another window.

Instead of capturing any window with an empty/matching title created by
the same process, the cropping capturer will capture any window created
by the same thread. While not foolproof, this heuristic seems to capture
menus/tooltips from the window of interest while excluding those from
other top-level windows in practice (assuming those were created by a
separate thread / independent message pump).

Bug: webrtc:10856
Change-Id: I2072c79da9e0158475b442a43b5b96d6ad307bc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148641
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#28824}
2019-08-10 03:40:17 +00:00
Niels Möller
b90d38a978 Delete unused Opus-specific methods of AudioCodingModule
Bug: None
Change-Id: Ib191e4beadf85cd57e765bc52d305e274e50a473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148400
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28815}
2019-08-09 07:06:36 +00:00
Niels Möller
5297cf368d Delete unused class MockTargetTransferRateObserver
Bug: None
Change-Id: I60e9dc05450207dfd572ae17a42cf1adaed4c1b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28813}
2019-08-09 06:15:06 +00:00
Gustaf Ullberg
940c2b5005 AEC3: Reduce level of log messages
This change reduces the level of several non-critical log messages in
order to reduce log spamming.

Bug: webrtc:8671
Change-Id: I6faae7a2ae4eeafd18c2770208485a75ad946e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148528
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28809}
2019-08-08 14:08:05 +00:00
Niels Möller
0d210ee2f6 Change return type of of ReceiveStatistics::Create to unique_ptr.
There are currently three overloads with different number of arguments,
and one of those return a raw pointer. This cl changes that to unique_ptr.
The transition plan is to update those downstream call sites that
currently require a raw pointer to use one of the other overloads.

Bug: webrtc:10679
Change-Id: I234605e99c04a59fbe6f478581ed8edd96a9b05a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28804}
2019-08-08 08:19:43 +00:00
Erik Språng
c2fe547eba Remove unused fallbacks in PacedSender
Bug: webrtc:10809
Change-Id: I322e5f0dbfb8648aee4f88d37b8a0938a48c0f3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148440
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28803}
2019-08-08 08:18:38 +00:00
Per Åhgren
eac47f7fae Removing unused fallback variant for the reverb computation
This CL removes a long unused fallback behavior for the reverb
computation.

Bug: webrtc:8671
Change-Id: I4b57795a9bb33769237858f40392881ee235653e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148520
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28802}
2019-08-08 08:00:38 +00:00
Alex Narest
5b5d97c938 Reland of "Reporting of decoding_codec_plc events""
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.

The new stat will not be reported unless it is GT 0.

Reporting of decoding_codec_plc events

Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
2019-08-07 18:41:46 +00:00
Oleh Prypin
b1686786e8 Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.

References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/
https://stackoverflow.com/a/2524673

Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
2019-08-07 13:36:05 +00:00
Niels Möller
12ebfa69ba Delete RtcpStatisticsCallback from ReceiveStatistics
Update VideoReceiveStream::GetStats to use
StreamStatistician::GetStatistics instead, similarly to the audio
receiver.

Bug: webrtc:10679
Change-Id: I8a701e8a8e921c87895424362dc83500737c916d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142233
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28793}
2019-08-07 13:33:55 +00:00
Danil Chapovalov
83bbe91398 Delete deprecated rtc_event_log header
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 10:58:17 +00:00
Chen Xing
e08648dc70 Add AbsoluteCaptureTime to RtpPacketInfo.
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.

Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
2019-08-07 10:12:56 +00:00
Niels Möller
75caef7a4b Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_
Unused since https://webrtc-review.googlesource.com/c/103821.

Bug: webrtc:8396
Change-Id: Ia83f02f16d6ea8c260ea765b41506f2691e035bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148072
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28788}
2019-08-07 09:53:22 +00:00
Niels Möller
c653172e74 Delete obsolete method AudioCodingModule::SetBitRate
Bug: None
Change-Id: I2291f7b4b46d269592eacad67a126010b750fac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148079
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28784}
2019-08-07 08:37:25 +00:00