Reason for revert:
Break performance bots.
Original issue's description:
> Fix off-by-one bugs in video_coding::PacketBuffer when the buffer is filled with a single frame.
>
> BUG=webrtc:8028
>
> Review-Url: https://codereview.webrtc.org/2993513002
> Cr-Commit-Position: refs/heads/master@{#19209}
> Committed: ee13e8919cTBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8028
Review-Url: https://codereview.webrtc.org/2990183002
Cr-Commit-Position: refs/heads/master@{#19211}
Add SetProcessParams method for configuring process settings (removes intermediate step of configuring settings via ProcessParams).
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2962293002
Cr-Commit-Position: refs/heads/master@{#19206}
This change adds constructors for all DesktopFrame inheritances to pass in
DesktopRect instead of DesktopSize.
Because the newly added constructors and DesktopFrame::top_left() function are
not actively used, this change should have no logic impact.
Bug: webrtc:7950
Change-Id: If78187865c991211dfc28d3723403ce6e6fe0290
Reviewed-on: https://chromium-review.googlesource.com/590508
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19204}
We missed to implement quality scaling in the original CL
https://codereview.webrtc.org/2977213002/. This CL implements it.
Note that the ObjC interface for scalingSettings is slightly different from the C++
interface in that we require explicit QP thresholds to turn quality scaling on, i.e.
we don't provide default values. I think this is more modular as we want to move
codec specific knowledge out from the WebRTC core. I would like to update the
C++ webrtc::VideoEncoder interface to do the same in another CL.
BUG=webrtc:7924
Review-Url: https://codereview.webrtc.org/2991123002
Cr-Commit-Position: refs/heads/master@{#19202}
If the input file name matches the "<name>-<params>.wav" pattern and <name> is a valid signal creator name, then <params> is parsed and used to create a new signal which is written in place of the missing file.
This CL only adds a pure tone creator. For instance, 'pure_tone-440_1000.wav' creates a pure tone at 440 Hz, 1000 ms long, mono, sampled at 48kHz.
This feature can be used to simplify the creation of common probe signals - no need to add external .wav files. Also, it will be exploited by a coming CL that adds a new evaluation score requiring the input signal to be a pure tone.
Additional minor fixes:
- apm_quality_assessment_unittest.py: command line arguments replaced to avoid that those for the unit test framework are passed
- simulation_unittest.py: invalid evaluation score name replaced
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2989823002
Cr-Commit-Position: refs/heads/master@{#19200}
Also change the loss rates to 5% and 1%, instead of 50%.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2950313002
Cr-Commit-Position: refs/heads/master@{#19199}
- render stream support, required to assess AEC;
- echo path simulation and input mixer, to generate echo and add it to the
speech signal;
- export engine: improved UI, switch to Pandas DataFrames;
- minor design improvements and needed adaptions.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2813883002
Cr-Commit-Position: refs/heads/master@{#19198}
rtcp_sender accepts nullptr as indication statistics shouldn't be used,
Other uses of NullReceiveStatistcs were already deleted.
BUG=webrtc:8016
Review-Url: https://codereview.webrtc.org/2988143002
Cr-Commit-Position: refs/heads/master@{#19197}
Added documentation of thread expectations for video tracks and sources to the API.
Originally landed as patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/.
Patchset 1 is the originall cl.
Patschet 2 is modified so that VideoTrackInterface::AddSink and RemoveSink have a default implementation.
BUG=none
Review-Url: https://codereview.webrtc.org/2989113002
Cr-Commit-Position: refs/heads/master@{#19195}
Broken by https://codereview.webrtc.org/2750783004/. Since samples are
two bytes each, only half of the buffer was being zeroed, leading to
garbage noise.
BUG=webrtc:7885,webrtc:7343
Change-Id: I46ecf90258b681ccdebbcfadd2e84ac6abadc9fe
Reviewed-on: https://chromium-review.googlesource.com/593092
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19194}
I am not sure memcmp is the right tool to compare two D3D11_TEXTURE2D_DESC
instances. So the staging texture may be recreated for each frame, which hurts
the performance.
Bug: webrtc:8046
Change-Id: I60a94f468599b23dec168de55c9bc8c787ab9b7d
Reviewed-on: https://chromium-review.googlesource.com/592088
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19193}
resolution_tracker_ should always represent the size of the DxgiFrame::frame_.
So it should not be actively reset.
Bug: webrtc:8045
Change-Id: I0b4d70ea69e4c2febfa369de50b555287c41fd99
Reviewed-on: https://chromium-review.googlesource.com/592248
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19192}
DxgiTexture now does not rely on a fixed resolution, so the ResolutionTracker
can be removed from it.
This change does not have logic impact, the upper component
(DxgiDuplicatorController) always reinitializes itself once the screen
resolution changes. And this check is also a legacy one: DxgiFrame now can take
care of the resolution change itself without needing to return false in
DxgiTexture.
Bug: webrtc:8044
Change-Id: I3ad9ce175f2bc9bf03b0a3985efa2681aa55d14b
Reviewed-on: https://chromium-review.googlesource.com/592247
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19191}
RTCEAGLVideoVideo ensureGLContext has been observed to fail because the
GL context is nil. This CL checks the GL context is non-nil in the ctor
instead.
BUG=b/62865840
Review-Url: https://codereview.webrtc.org/2991863002
Cr-Commit-Position: refs/heads/master@{#19189}
ResolutionChangeDetector now does not update its internal state. There is no
impact because Reset() is always actively called.
So this change renames ResolutionChangeDetector to ResolutionTracker, and rename
the IsChanged() function into SetResolution(), which returns true if a
replacement happened. Internally it always records the latest DesktopSize.
Customers of this class can still use SetResolution() function to check whether
a DesktopSize change happened.
Bug: webrtc:8038
Change-Id: I6d25f3dd2d0567219a82b6688bf3e08560c8b0af
Reviewed-on: https://chromium-review.googlesource.com/587405
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19184}
MockRtpPacketSink has three identical implementations now, so time to move it to its own file.
BUG=None
Review-Url: https://codereview.webrtc.org/2988853002
Cr-Commit-Position: refs/heads/master@{#19183}
Rtcp sender now take smaller interface making it possible to simplify the fake
BUG=webrtc:8016
Review-Url: https://codereview.webrtc.org/2984283002
Cr-Commit-Position: refs/heads/master@{#19181}
Previously, the matrix in VideoFrame was used to crop and scale the
frame. This caused complications because webrtc::VideoFrame doesn't
include a matrix. cropAndScale method is added to VideoBuffer class for
cropping and scaling instead.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/2990583002
Cr-Commit-Position: refs/heads/master@{#19179}
I don't think this line could never conceivably fail - if the ctor has reached that point, the object fit in memory, and its members have all been allocated legal memory addresses, none of which may be 0x00.
BUG=None
Review-Url: https://codereview.webrtc.org/2989813002
Cr-Commit-Position: refs/heads/master@{#19176}
This will result in the ipsec interfaces being prioritized below Wi-Fi
and cell interfaces. This makes the most difference when we hit the
default limit for IPv6 interfaces (5), and there are lots of ipsec
interfaces for whatever reason, resulting in the "real" interfaces that
would actually succeed not being used. See the linked bug 7703.
BUG=webrtc:7703, webrtc:3149
Review-Url: https://codereview.webrtc.org/2985133002
Cr-Commit-Position: refs/heads/master@{#19175}
Chromium has adopted Opus 1.2.1 which allows 120ms frame encoding. It
is time to turn on the switch for building WebRTC with this feature.
Bug: webrtc:8042
TBR: kjellander@webrtc.org
Change-Id: I644b47cfb56f835695ef1263741cda6e3ee3d862
Reviewed-on: https://chromium-review.googlesource.com/586725
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Felicia Lim <flim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19173}
Incompatible upstream changes will be detected in Chromium roll
instead of breaking presubmit
find_depot_tools is already aware of this, see
cea92c51ac%5E%21
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2986003002
Cr-Commit-Position: refs/heads/master@{#19165}
