1353 Commits

Author SHA1 Message Date
Ivo Creusen
c7f09ad2e0 NetEq fix for repeated audio issue.
This CL implements a fix behind a field trial for a NetEq issue. NetEq restarts audio too quickly after a buffer underrun, which can quickly lead to another underrun in some circumstances. The fix changes NetEq's behavior to wait with restarting playback until sufficient audio is buffered.

Bug: webrtc:9289
Change-Id: I5968c9478ce8d84caf77f00b8d0a39156b47fc8d
Reviewed-on: https://webrtc-review.googlesource.com/77423
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23347}
2018-05-22 12:57:58 +00:00
Kári Tristan Helgason
169005d8c1 Move VideoCodecTest configuration classes to api/test.
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.

This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.

Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 12:14:38 +00:00
Henrik Lundin
6dc82e8f8b NetEq: Change NetEq's ramp-up behavior after expansions
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.

This change breaks bit-exactness, but careful listening could not reveal
an audible difference.

This change is a part of a larger refactoring of NetEq's PLC code.

Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
2018-05-22 09:38:28 +00:00
Ying Wang
7a84fcf47a Prevent potential buffer overflow in UlpfecReceiver
Bug: chromium:841962
Change-Id: I5ef0341a5fffe6b6204f5b2edbaec2d389a56964
Reviewed-on: https://webrtc-review.googlesource.com/77420
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23341}
2018-05-22 09:32:18 +00:00
Niels Möller
401d07690b Delete deprecated VideoDecoder::Decode method
Follow up to https://webrtc-review.googlesource.com/c/src/+/39511,
which introduced a new Decode method, without the
RTPFragmentationHeader argument, and deprecated the old method.

Bug: webrtc:6471
Change-Id: Icd3c536ebedd4e3c2d57fdb4d6e078d6ff1de5b6
Reviewed-on: https://webrtc-review.googlesource.com/75180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23339}
2018-05-22 08:17:03 +00:00
Gustaf Ullberg
41c11e4cad AEC3: Rounding of estimated call skew
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).

Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
2018-05-22 08:15:58 +00:00
Niels Möller
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
Gustaf Ullberg
b9fc6508c0 Add min and max allowed bitrate in Opus bitrate tests
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.

Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
2018-05-21 16:41:35 +00:00
Jesús de Vicente Peña
666becad58 AEC3: ERLE improvements
The ERLE computation was improved by two means:
- The update function was always called and just parts of the internal code reacts to the converged filter flag
- When computing the ERLE, the ratio of energies is now computed using more points and, therefore, a more robust estimation is achieved.

Bug: webrtc:9284
Change-Id: Ie4f871f19cfad1a13741352ddd7b0a27ad6c3fb6
Reviewed-on: https://webrtc-review.googlesource.com/77767
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23329}
2018-05-21 15:11:06 +00:00
Henrik Lundin
9024da84c9 NetEq: Fixing an overflow bug in expand.cc
The overflow currently does not cause any problems, but it has been
found that it can cause crashes after a refactoring that is coming in
the near future.

Bug: webrtc:9180
Change-Id: Ia2c4e545c062c4f8ad13cbc47b8796c6e8a4e906
Reviewed-on: https://webrtc-review.googlesource.com/77667
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23327}
2018-05-21 13:39:25 +00:00
henrika
9ab6eb738a Minor namespace change for CoreAudioUtility
NOTRY=TRUE

TBR: kwiberg@webrtc.org
Bug: webrtc:9265
Change-Id: Ic40634eb5258739ef06becd5db7a70a1e31d29e3
Reviewed-on: https://webrtc-review.googlesource.com/78020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23323}
2018-05-21 12:28:25 +00:00
Gustaf Ullberg
6633d41bb0 Reland "Update expected bitrate in Opus tests"
This is a reland of 79ded653fee7183d5c0d94c5addf570bcfb29c9e

Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I6bfcd1c5e1d5298543024a0faa6a695026434df3
Reviewed-on: https://webrtc-review.googlesource.com/77980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23318}
2018-05-21 08:13:05 +00:00
Sergey Silkin
d902d58b0a Framerate controller for VP9 screen sharing.
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.

Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
2018-05-19 07:14:48 +00:00
Sebastian Jansson
dfce03af6e Allows injection of network controller factory into peer connection factory.
Bug: webrtc:9155
Change-Id: I0a17024042f154297aba20f5d2dc766feb27f3f7
Reviewed-on: https://webrtc-review.googlesource.com/73123
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23313}
2018-05-18 17:07:16 +00:00
Sergey Silkin
be71a1ee08 Replace VP9 screen sharing.
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.

Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
2018-05-18 15:11:46 +00:00
Stefan Holmer
812ceafb5a Ensure render time is zero when playout delay is zero so that minimal latency in the render pipeline is ensured.
Bug: webrtc:9135
Change-Id: Id9ae8ec59536808ba8923c73dd46abfe3fa6fe79
Reviewed-on: https://webrtc-review.googlesource.com/75600
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23309}
2018-05-18 14:47:26 +00:00
Gustaf Ullberg
6bf5a0d5b6 AEC3: High-pass filter delay estimator signals
This CL applies a high pass filter to the delay estimator signals which
improves the adaptation of the matched filters in noisy environments.
This results in faster delay estimation.

Bug: webrtc:9288
Change-Id: I8ffe5442eab7ac2f10a7ba236b08a0f07ec90645
Reviewed-on: https://webrtc-review.googlesource.com/77725
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23308}
2018-05-18 14:33:26 +00:00
Gustaf Ullberg
77995e744b Revert "Update expected bitrate in Opus tests"
This reverts commit 79ded653fee7183d5c0d94c5addf570bcfb29c9e.

Reason for revert: Different repos have different Opus

Original change's description:
> Update expected bitrate in Opus tests
> 
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
> 
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}

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

Change-Id: I3c18db2d6052c4049d836c3e595b00189aebcbc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9280
Reviewed-on: https://webrtc-review.googlesource.com/77800
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23307}
2018-05-18 14:27:36 +00:00
Gustaf Ullberg
79ded653fe Update expected bitrate in Opus tests
Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
CL re-enables recently disabled unittests and updates the expected bitrates.

Bug: webrtc:9280
Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
Reviewed-on: https://webrtc-review.googlesource.com/77766
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
2018-05-18 13:45:06 +00:00
Per Åhgren
2d9a3b1aba Increasing the API call skew hysteresis limit in AEC3
This CL increases the allowed variations in the API call skew limit in
AEC3.

Bug: webrtc:9283,chromium:888042
Change-Id: Ib5e784c6f3dcf1bf3a2cbfe2b1559953db9227a8
Reviewed-on: https://webrtc-review.googlesource.com/77430
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23305}
2018-05-18 13:39:26 +00:00
Niels Möller
0a8f43580f Move VideoEncoderConfig from call/ to api/.
Bug: webrtc:8830
Change-Id: I42abd45bff9a70fe00733424b34874925c523dc8
Reviewed-on: https://webrtc-review.googlesource.com/77683
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23303}
2018-05-18 12:58:16 +00:00
henrika
7e6fcea7de Ports CoreAudioUtil from Chrome to WebRTC.
See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34
for details.

Bug: webrtc:9265
Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb
Reviewed-on: https://webrtc-review.googlesource.com/76900
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:25:26 +00:00
Anders Carlsson
7eb8e9fd7b Add RegisterExternalDecoder in VideoCodingModule.
In preparation for landing https://webrtc-review.googlesource.com/c/src/+/72441
a downstream project that uses the VideoCodingModule needs to be able to
inject a decoder object created from the outside, just like how encoders
are possible to inject.

Bug: webrtc:7925
Change-Id: Ibaeffda55f84410436d79f75730e7352e298b9f0
Reviewed-on: https://webrtc-review.googlesource.com/77160
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23297}
2018-05-18 09:43:26 +00:00
Niels Möller
c948fe62fd Delete unneeded includes of call/video_config.h.
Bug: webrtc:8830
Change-Id: I6114b47e5524a6d2450108388236478b1ceafb67
Reviewed-on: https://webrtc-review.googlesource.com/77425
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23295}
2018-05-18 09:00:56 +00:00
Mirko Bonadei
638edfc88c Skipping some Opus tests to let the new roll flow.
In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].

They will be re-enabled and fixed as soon as the new Opus revision is
rolled.

[1] - https://chromium-review.googlesource.com/1061499

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
2018-05-18 07:58:46 +00:00
Niels Möller
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
Per Kjellander
dd3eae5f94 Revert "Configure and use max bitrate to limit the AIMD controller estimates."
This reverts commit 18d7c7ea7e56444d6d7e6c8fb95b5f426fd7b953.

Reason for revert: 
This seems to cause the auto roller to Chrome to fail on Linux and Mac on the browsertest
WebRtcSimulcastBrowserTest.TestVgaReturnsTwoSimulcastStreams

https://chromium-review.googlesource.com/c/chromium/src/+/1064736


Original change's description:
> Configure and use max bitrate to limit the AIMD controller estimates.
> 
> Bug: webrtc:9275
> Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
> Reviewed-on: https://webrtc-review.googlesource.com/77081
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23287}

TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: I8ed827ab6b2f7d2b70b9889e5a88701bfb974d35
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9275
Reviewed-on: https://webrtc-review.googlesource.com/77660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23291}
2018-05-18 07:12:26 +00:00
braveyao
90394a4db3 Reland "[desktopCapture] Unify the position info in DIP coordinates on Mac."
This is a reland of 89653d5db46419d2a80898635cb27fed64898db2

Original change's description:
> [desktopCapture] Unify the position info in DIP coordinates on Mac.
> 
> On OSX, the logical(DIP) and physical coordinates are used mixingly.
> For example, the captured image has its size in physical pixels(2x) and
> location in logical(DIP) pixels. Same to the cursor position. This
> causes trouble when we check the relative position of image and cursor
> when there are multiple monitors with different DIP setting connected.
> 
> This cl proposed a solution to use DIP pixel for any location info,
> i.e. top-left of a frame and cursor position. Also propose a method to
> get the current scale factor of a window across multiple monitors. And
> save the current scale factor in DPI of the capture frame.
> Then we can check relative position of cursor and frame correctly
> in DIP pixel and compose them in physical pixel.
> 
> Bug: webrtc:9178
> Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
> Reviewed-on: https://webrtc-review.googlesource.com/71621
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Reviewed-by: Zijie He <zijiehe@chromium.org>
> Commit-Queue: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23263}

Bug: webrtc:9178
Change-Id: I97d9150f7b9a4ed6671733b75613ea9c315d5c1d
Reviewed-on: https://webrtc-review.googlesource.com/77481
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23289}
2018-05-17 18:45:42 +00:00
Bjorn Terelius
18d7c7ea7e Configure and use max bitrate to limit the AIMD controller estimates.
Bug: webrtc:9275
Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
Reviewed-on: https://webrtc-review.googlesource.com/77081
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23287}
2018-05-17 16:07:42 +00:00
philipel
09133af36f Check number of nalus in packet before checking nalu types.
Bug: chromium:840536
Change-Id: Ia4dcf322ad6290691fd01b58fb02cd868714c92e
Reviewed-on: https://webrtc-review.googlesource.com/77121
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23283}
2018-05-17 12:52:11 +00:00
Erik Språng
8b7ca4abb2 Make packet router send padding on rtp module that last sent media.
Currently we prefer the last added rtp module that supports rtx, and
assume this is the HD stream.

If we suffer a network degradation and stop sending HD, the current
behavior will trigger RTX padding on an inactive stream, which is not
very useful.

With this change, we will prefer the rtp module that last sent media,
which will spread the load a bit across active media streams, but will
be biased toward the one with highest packet rate.

Bug: webrtc:8975
Change-Id: Id52865ccd5263722c66d327b8c80457f63b90385
Reviewed-on: https://webrtc-review.googlesource.com/77360
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23281}
2018-05-17 12:21:11 +00:00
Per Kjellander
e1c3c01a90 Revert "[desktopCapture] Unify the position info in DIP coordinates on Mac."
This reverts commit 89653d5db46419d2a80898635cb27fed64898db2.

Reason for revert: 

Tentatively revert since I believe this break remoting unittests on Asan/Tsan

https://chromium-review.googlesource.com/c/chromium/src/+/1063330

https://chromium-swarm.appspot.com/task?id=3d8692bedcc85c10&refresh=10&show_raw=1


Original change's description:
> [desktopCapture] Unify the position info in DIP coordinates on Mac.
> 
> On OSX, the logical(DIP) and physical coordinates are used mixingly.
> For example, the captured image has its size in physical pixels(2x) and
> location in logical(DIP) pixels. Same to the cursor position. This
> causes trouble when we check the relative position of image and cursor
> when there are multiple monitors with different DIP setting connected.
> 
> This cl proposed a solution to use DIP pixel for any location info,
> i.e. top-left of a frame and cursor position. Also propose a method to
> get the current scale factor of a window across multiple monitors. And
> save the current scale factor in DPI of the capture frame.
> Then we can check relative position of cursor and frame correctly
> in DIP pixel and compose them in physical pixel.
> 
> Bug: webrtc:9178
> Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
> Reviewed-on: https://webrtc-review.googlesource.com/71621
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Reviewed-by: Zijie He <zijiehe@chromium.org>
> Commit-Queue: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23263}

TBR=zijiehe@chromium.org,jamiewalch@chromium.org,perkj@webrtc.org,braveyao@webrtc.org

Change-Id: Ica02365925623e21b256d20a21b5625e7ed6f49b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9178
Reviewed-on: https://webrtc-review.googlesource.com/77461
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23280}
2018-05-17 12:11:22 +00:00
Erik Språng
eeaa8f929c Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
Bug: webrtc:9271
Change-Id: Ic7415830588bef9d87bab92943460207890dada6
Reviewed-on: https://webrtc-review.googlesource.com/76960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 11:22:56 +00:00
philipel
d5addcaf1e Add philipel to modules/video_coding/OWNERS.
Bug: None
Change-Id: I982ac84a58aff27a6f8f9cf50005ac732f540785
Reviewed-on: https://webrtc-review.googlesource.com/77362
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23274}
2018-05-17 10:14:31 +00:00
Kári Tristan Helgason
8eeda499f7 Enable more VideoCodecTests on iOS.
Bug: webrtc:4755
Change-Id: I403834dbe04cc3899847eb10e5595a24e6001507
Reviewed-on: https://webrtc-review.googlesource.com/76602
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23273}
2018-05-17 09:13:31 +00:00
Per Åhgren
90e3fbdd37 Activating the AEC3 audibility improvements functionality
This CL turns on the previously implemented AEC3 audibility
improvements, which before has been off by default.

Bug: webrtc:9193,chromium:836790
Change-Id: Ibcd057ba5dd002718d62fd83db33d01d9563b8ea
Reviewed-on: https://webrtc-review.googlesource.com/77123
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23265}
2018-05-16 16:47:16 +00:00
braveyao
89653d5db4 [desktopCapture] Unify the position info in DIP coordinates on Mac.
On OSX, the logical(DIP) and physical coordinates are used mixingly.
For example, the captured image has its size in physical pixels(2x) and
location in logical(DIP) pixels. Same to the cursor position. This
causes trouble when we check the relative position of image and cursor
when there are multiple monitors with different DIP setting connected.

This cl proposed a solution to use DIP pixel for any location info,
i.e. top-left of a frame and cursor position. Also propose a method to
get the current scale factor of a window across multiple monitors. And
save the current scale factor in DPI of the capture frame.
Then we can check relative position of cursor and frame correctly
in DIP pixel and compose them in physical pixel.

Bug: webrtc:9178
Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
Reviewed-on: https://webrtc-review.googlesource.com/71621
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23263}
2018-05-16 16:44:01 +00:00
Sebastian Jansson
a8bf169906 Adds BandwidthSampler for BBR.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.

Bug: webrtc:8415
Change-Id: Ic2dc4394dc9923e5109ffa5f146c23b527f0c395
Reviewed-on: https://webrtc-review.googlesource.com/76582
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23262}
2018-05-16 16:11:21 +00:00
Rasmus Brandt
8dd4db49e2 Create/destroying codecs on task queue + switch to TaskQueueForTest.
After https://webrtc-review.googlesource.com/c/src/+/70740, we are
creating/destroying the codecs on a task queue in the VideoStreamEncoder. This
CL updates the VideoCodecTest to do the same.

Also, this CL switches from manually Wait()'ing on the task queue to using
TaskQueueForTest::SendTask.

Bug: None
Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e
Reviewed-on: https://webrtc-review.googlesource.com/76800
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 08:15:23 +00:00
Niels Möller
59130a11b8 Delete deprecated version of VideoCodecInitializer::SetupCodec.
A followup to https://webrtc-review.googlesource.com/71380 and
https://webrtc-review.googlesource.com/69986, deleting the
nack_enabled flag.

Bug: webrtc:8830
Change-Id: Ie53f7a1e131be5347936ed44cee7167295026d57
Reviewed-on: https://webrtc-review.googlesource.com/76760
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23256}
2018-05-16 08:03:24 +00:00
Mirko Bonadei
d7624f2d90 Removing clang-cl warning suppression flags in video_capture.
After landing https://webrtc-review.googlesource.com/76720 and
https://webrtc-review.googlesource.com/76722 it is possible to remove
clang-cl only warning suppression flags.

These were needed because winsdk_samples was included using include_dirs
(-I) instead of -imsvc (which is similar to -isystem in GCC and clang).

Bug: webrtc:9251
Change-Id: I2bc19f0ba552342c49a29c07e4fb84d439003dfb
Reviewed-on: https://webrtc-review.googlesource.com/76781
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23252}
2018-05-15 17:33:32 +00:00
Alessio Bazzica
2f1e6d4920 AGC2 RNN VAD: Polishing.
- Code clean: exploiting the recently added ArrayView ctor for
  std::array
- Pitch search internal unit test: long const arrays moved to
  a resource file
- Minor changes

Bug: webrtc:9076
Change-Id: Iaf30753f2498b4568860d72e0b81f5351235692f
TBR: aleloi@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/76920
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23248}
2018-05-15 16:41:02 +00:00
Alex Loiko
1aec594018 Merge :audio_processing and :aec_dump_interface.
Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502

':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.

Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.

NOTRY=true # want this in and 'ios_api_framework' seems stuck.

Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
2018-05-15 14:22:52 +00:00
Sam Zackrisson
ae93f0412a Make an energy computation not overflow in iLBC PLC
The current implementation carefully shifts down the energy so as not to overflow.
The fuzzer audio_decoder_ilbc_fuzzer found an integer overflow anyway.
The energy is only used for a threshold check.

This fix stops the energy computation when the threshold is reached, before it can overflow.

Bug: chromium:837922
Change-Id: I45e84d2d271a37e6476b08433a2cbd5a8f6e6f26
Reviewed-on: https://webrtc-review.googlesource.com/76122
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23242}
2018-05-15 13:01:42 +00:00
Rasmus Brandt
8694f29b30 Rename VideoProcessorIntegrationTest -> VideoCodecTest.
This CL simply renames the test cases that were not renamed in
prior CLs.

Bug: None
Change-Id: If616eb823e1453bc92ba037722b77a219d54409c
Reviewed-on: https://webrtc-review.googlesource.com/76780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23240}
2018-05-15 11:28:21 +00:00
Peter Hanspers
8d95e3b211 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
2018-05-15 10:33:01 +00:00
Alex Loiko
73ec01977b Add RuntimeSettings to CustomProcessing.
CustomProcessing is the interface to injectable audio processing
submodules to AudioProcessing. This CL makes it possible to set
runtime settings on the injected render processing component.

Note that the current runtime setting handling happens on the capture
thread. Therefore, we add another SwapQueue to communicate with the
render thread.

Bug: webrtc:9138, webrtc:9262
Change-Id: I665ce2d83a2b35ca8b25cca813d2cef7bd0ba911
Reviewed-on: https://webrtc-review.googlesource.com/76123
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23236}
2018-05-15 10:03:25 +00:00
Alessio Bazzica
beb1d34729 AGC2 RNN VAD: Feature extraction.
This CL finalizes the feature extraction part for the RNN VAD adding
a class that combines a high-pass filter, LP residual computation,
pitch estimation and spectral features extraction.

This CL also includes a minor refactoring of the pitch estimation
library.

Bug: webrtc:9076
Change-Id: I918b9e143bc6dd2bf508a891446067258a68a777
Reviewed-on: https://webrtc-review.googlesource.com/75504
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23235}
2018-05-15 10:02:20 +00:00
Rasmus Brandt
a7e4844298 Move H.264 codec test activation directly behind build flag.
Bug: None
Change-Id: I5dd366aba16b440a8e99826e116db0d6b68bf844
Reviewed-on: https://webrtc-review.googlesource.com/76621
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23231}
2018-05-15 08:00:00 +00:00
Sebastian Jansson
30bbbe8e79 Adds PacketNumberIndexedQueue for BBR.
This prepares for making the BBR implementation more identical to the
implementation in Quic, to ensure that results are comparable.

Bug: webrtc:8415
Change-Id: Icc66b748e52efb80b5c3ade85f83de6b224a4bd2
Reviewed-on: https://webrtc-review.googlesource.com/76581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23227}
2018-05-15 07:09:50 +00:00