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}
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}
CreateCroppedDesktopFrame() does not need to create a CroppedDesktopFrame if the
size won't change.
Bug: webrtc:8039
Change-Id: Ie6789a4b473b69bced94c4a25a68f1da6bb3510e
Reviewed-on: https://chromium-review.googlesource.com/587808
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19163}
This allows an application to easily override the default limit
(currently 5).
Also adding a test that covers more of the
PeerConnection<->PortAllocator interaction.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2985653003
Cr-Commit-Position: refs/heads/master@{#19160}
Relanding because the broken chromium test has been fixed:
https://chromium-review.googlesource.com/582196
This CL moves the responsibility for restricting the number of IPv6
interfaces used for ICE to BasicPortAllocator. This is the right place
to do it in the first place; it's where all the rest of the filtering
occurs. And NetworkManager shouldn't need to know about ICE limitations;
only the ICE classes should.
Part of the reason I'm doing this is that I want to add a
"max_ipv6_networks" API to RTCConfiguration, so that applications can
override the default easily (see linked bug). But that means that
PeerConnection would need to be able to call "set_max_ipv6_networks" on
the underlying object that does the filtering, and that method isn't
available on the "NetworkManager" base class. So rather than adding
another method to a place it doesn't belong, I'm moving it to the place
it does belong.
In the process, I noticed that "CompareNetworks" is inconsistent with
"SortNetworks"; the former orders interfaces alphabetically, and the
latter reverse-alphabetically. I believe this was unintentional, and
results in undesirable behavior (like "eth1" being preferred over
"eth0"), so I'm fixing it and adding a test.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Original-Commit-Position: refs/heads/master@{#19112}
Committed: ad9561404c
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Commit-Position: refs/heads/master@{#19159}
This change returns translated position in the newly added overload
MouseCursorMonitor::Callback::OnMouseCursorPosition(DesktopVector) callback.
Meanwhile it also reduces the duplicate logic in Windows capturer
implementations. So except for the deprecated logic in MouseCursorMonitorWin,
all GetSystemMetrics() function calls are merged into GetScreenRect(),
GetFullscreenRect() and GetFullscreenTopLeft() functions.
Bug: webrtc:7950
Change-Id: Ic2a85a80b6947367bdd20d8f96f11e0f5c269006
Reviewed-on: https://chromium-review.googlesource.com/581951
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19157}
When you create multiple "PeerConnectionFactory"s, they end up using
the same NetworkMonitor singleton. But the second one's
"AndroidNetworkMonitor" class (in C++) wasn't getting the expected
network list update, and as a result it wasn't binding sockets to
networks successfully, acting as if the networks didn't exist.
The solution is just to move "updateActiveNetworkList" to
"startMonitoring". This CL also does some other minor
cleanup/refactoring, and fixes a more corner-casey issue where, if the
first PeerConnection is destroyed, the second one would stop receiving
network updates.
BUG=webrtc:7946
Review-Url: https://codereview.webrtc.org/2990693002
Cr-Commit-Position: refs/heads/master@{#19156}
There is already an Unwrapper in webrtc/modules/include/module_common_types.h,
but we reimplemented it in sequence_number_util.h for a few reasons:
- Such a class belongs in sequence_number_util.h.
- It is a cleaner implementation since we can use the rest of
sequence_number_util.h functionality.
- You can choose at which number the unwrapped sequence should start,
which is used to avoid the edge case when a backward wrap can happen
as the first few numbers are unwrapped.
- This unwrapper can unwrap numbers that does not wrap 8/16/32 bits.
BUG=None
Review-Url: https://codereview.webrtc.org/2977603002
Cr-Commit-Position: refs/heads/master@{#19154}
This CL was modified from work of sharifferdous@ (intern supervised by lliuu@)
BUG=webrtc:7389
Review-Url: https://codereview.webrtc.org/2987723002
Cr-Commit-Position: refs/heads/master@{#19146}
