VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.
Review URL: https://codereview.webrtc.org/1532133002
Cr-Commit-Position: refs/heads/master@{#11108}
1. It signals network changed events whenever there are more than one IP address in a network.
2. It does not signal network changed events if a network disconnects and connects again.
Also changed DumpNetworks for better debugging.
BUG=webrtc:5096
Review URL: https://codereview.webrtc.org/1421433003
Cr-Commit-Position: refs/heads/master@{#11107}
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.
No functional changes.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1537273003
Cr-Commit-Position: refs/heads/master@{#11101}
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
Currently, FFT is performance when AEC buffers farend signal. This has some drawbacks
1. memory inefficiency: two ring buffers are needed;
2. computation inefficiency: if ringbuffer gets wrapped around, some FFT computation will be wasted;
3. accessibility: the main AEC function looses accessibility to the time-domain signal.
Therefore, this CL tries to buffer time domain data, which is buffered any way if a debugging macro is defined, and calculate the FFTs where they are actually used.
BUG=
Review URL: https://codereview.webrtc.org/1512573003
Cr-Commit-Position: refs/heads/master@{#11091}
Add the --quiet flag to the download_from_google_storage runhooks
step to prevent it from spamming the console when all the files
are already downloaded.
NOTRY=True
Review URL: https://codereview.webrtc.org/1527713003
Cr-Commit-Position: refs/heads/master@{#11090}
This patch removes StringToIP() methods as fixes the TODO there and
there are no callers at the moment for these methods.
BUG=None
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1535993002
Cr-Commit-Position: refs/heads/master@{#11088}
Incorrect argument order, also added unittest which should've been there
in the first place.
Also renames AtomicLoadPtr to AcquireLoadPtr to match non-ptr version.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1537923003 .
Cr-Commit-Position: refs/heads/master@{#11086}
This should solve a problem discovered when converting from GYP to
other project formats, where the source files weren't included correctly
for each platform.
Two other targets in WebRTC have similar source files, which are correctly
generated for each platform:
* video_render_module_internal_impl
* video_capture_module_internal_impl
They both list the sources as it's changed to in this CL.
NOTRY=True
Review URL: https://codereview.webrtc.org/1536923003
Cr-Commit-Position: refs/heads/master@{#11083}
This removes a dependency on Chromium's build/build_config.h
(which is not allowed).
The added defines are identical to the ones in build/build_config.h.
NOTRY=True
Review URL: https://codereview.webrtc.org/1532333002
Cr-Commit-Position: refs/heads/master@{#11082}
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
There is an issue in PhysicalSocket::Accept where the flag to continue
listening is not set in "enabled_events_" if "accept" returns an error.
This CL fixes this (initial idea by silviu.cpp@gmail.com).
BUG=webrtc:2030
Review URL: https://codereview.webrtc.org/1452903006
Cr-Commit-Position: refs/heads/master@{#11080}
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.
We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.
BUG=568734
Review URL: https://codereview.webrtc.org/1532543003
Cr-Commit-Position: refs/heads/master@{#11075}
At speed 8, vp9 on ARM is currently ~2x times slower than vp8 on ARM (speed -12).
Update some parameters in videoprocessor_integrationtest.cc
to make tests pass on android (which uses the new speed setting).
TBR=stefan@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1526973004 .
Cr-Commit-Position: refs/heads/master@{#11072}
Also removes listing of targets in webrtc_fuzzers which is very prone to
not being up to date. They're not required for ClusterFuzz integration
or building locally. This also means that adding fuzzers won't require
approval outside the fuzzers directory.
BUG=webrtc:4771
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1518973003 .
Cr-Commit-Position: refs/heads/master@{#11067}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
