I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2963273002
Cr-Commit-Position: refs/heads/master@{#18856}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
Due to the implementation of the analog AGC in the audio
processing module, the detection for the analog gain done in AEC3
fails on some platforms where there is no analog gain to control.
This CL removes that functionality until the AGC behavior has
been corrected.
Bug: webrtc:7910, chromium:738322
Change-Id: Ibdbe1e02252387dfd94b36ba7471f5c56ae27f48
Reviewed-on: https://chromium-review.googlesource.com/556040
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18850}
This CL loosens the requirement on the API jitter in APM
that can be tolerated without affecting the AEC3 performance.
BUG=webrtc:7911,chromium:738323
Review-Url: https://codereview.webrtc.org/2967493004
Cr-Commit-Position: refs/heads/master@{#18849}
Leaving compatibility script in webrtc/tools/compare_videos.py to
avoid breaking our video quality tests in Chromium.
Forwarding GN targets are left in webrtc/tools/BUILD.gn.
BUG=webrtc:7855
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2965593002
Cr-Commit-Position: refs/heads/master@{#18848}
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.
BUG=webrtc:3411
Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
Linking external ObjC / Swift apps fails when the app code is using any
of the new frame-buffer classes RTCI420Buffer, RTCMutableI420Buffer, or
RTCCVPixelBuffer. To fix, we need to add the appropriate attribute to
the classes (e.g. using the RTC_EXPORT macro).
BUG=None
Review-Url: https://codereview.webrtc.org/2961293002
Cr-Commit-Position: refs/heads/master@{#18840}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Rewrite AsyncResolver to use PlatformThread directly, not
SignalThread, and update includes of peerconnection client to not
depend on signalthread.h.
BUG=webrtc:6424,webrtc:7723
Review-Url: https://codereview.webrtc.org/2915253002
Cr-Commit-Position: refs/heads/master@{#18833}
I intended to do this originally, but just forgot to pass the thread
with the virtual socket server into OrtcFactory::Create...
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2967453002
Cr-Commit-Position: refs/heads/master@{#18831}
Remove video codec settings from CodecParams (and rename to ProcessParams).
Removes intermediate step of configuring video settings via CodecParams.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2956243002
Cr-Commit-Position: refs/heads/master@{#18830}
magjed has written most of the code in this folder.
NOTRY=TRUE
Bug: None
Change-Id: I786261d4407f38de612f5fae12b9abde4594bac2
Reviewed-on: https://chromium-review.googlesource.com/550095
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18829}
Reason for revert:
Fix RtpStreamReceiver to not recover RTX packets with incorrect SSRC.
Original issue's description:
> Revert of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #5 id:120001 of https://codereview.webrtc.org/2893293003/ )
>
> Reason for revert:
> Breaks fuzzer.
>
> Original issue's description:
> > Only compare sequence numbers from the same SSRC in ForwardErrorCorrection.
> >
> > Prior to this CL, the ForwardErrorCorrection state would be reset whenever
> > the difference in sequence numbers of the last recovered media packet
> > and the new packet (media or FEC) was too large. This comparison did not
> > take into account that FlexFEC uses a different SSRC for the FEC packets,
> > meaning that the the state would be reset very frequently when FlexFEC
> > is used. This should not have led to any major problems, except for a
> > decreased decoding efficiency.
> >
> > This CL verifies that whenever we compare sequence numbers in
> > ForwardErrorCorrection, they do indeed belong to the same SSRC.
> >
> > BUG=webrtc:5654
> >
> > Review-Url: https://codereview.webrtc.org/2893293003
> > Cr-Commit-Position: refs/heads/master@{#18399}
> > Committed: 1476a9d789
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2919313005
> Cr-Commit-Position: refs/heads/master@{#18446}
> Committed: 92732ecc5cR=stefan@webrtc.org
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2918333002
Cr-Commit-Position: refs/heads/master@{#18827}
Don't disable fec for packets without timing frames extension
even if they are marked as belonging to timing frames.
BUG=webrtc:7894
Review-Url: https://codereview.webrtc.org/2956263002
Cr-Commit-Position: refs/heads/master@{#18826}
In most cases we can just return a frame referencing the buffer
returned by the decoder.
Bug: webrtc:7760
Change-Id: I0b42ab9662b39149e42a3c83adfd38a9d80e0e30
Reviewed-on: https://chromium-review.googlesource.com/544299
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18824}
Some .proto files have newline at the end. This CL levels all our .proto
files. A presubmit check will follow.
NOTRY=True
TBR=minyue@webrtc.org
Bug: None
Change-Id: I988fe94c31abf91c85a45b564c488329d677b958
Reviewed-on: https://chromium-review.googlesource.com/552137
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18823}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
Will reland in two different commits to preserve git blame history.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}
Reorganized the targets in webrtc/sdk/BUILD.gn so that the applications which use
WebRTC DataChannel only can depend on the "peerconnection_factory_no_media"
instead of "rtc_sdk_objc" to reduce the binary size.
Provided a no-media implementation of RTCPeerConnectionFactory using the macro
"HAVE_NO_MEDIA".
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2944643002
Cr-Commit-Position: refs/heads/master@{#18819}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
Was only working when the nonstandard "renomination" extension to ICE
is enabled, which chromium doesn't use.
BUG=chromium:734094
Review-Url: https://codereview.webrtc.org/2957303002
Cr-Commit-Position: refs/heads/master@{#18814}
Rationale:
1. sinks_ is not properly differentiated from rsid_sinks_.
2. Consistency with RtcpDemuxer.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2958283002
Cr-Commit-Position: refs/heads/master@{#18813}
Having members in a class which only exist when certain compliation flags are turned on (unless relating to the target platform) means that those flags must be the same when compiling the module as when including its headers from other modules. This means that code outside of WebRTC runs the risk of misjudging the size of an rtc::CriticalSection, or any class which has an rtc::CriticalSection as a member. (This rule is applied recursively.) If a mismatch occurs, memory corruption is likely.
Having discussed this a bit, we have decided that the simplest solution is probably the best - always define those members, even when compilation flags (namely, CS_DEBUG_CHECKS) render it unused.
BUG=webrtc:7867
Review-Url: https://codereview.webrtc.org/2957753002
Cr-Commit-Position: refs/heads/master@{#18811}
It allows to use rtc::Buffer in templates that expect std container,
e.g. it can now be used as ::testing::ElementsAreArray parameter
Bug: None
Change-Id: I97d7ffb13393d02850ddb213f7a1d01129b10b05
Reviewed-on: https://chromium-review.googlesource.com/539635
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18809}
This is in preparation of upgrading to libsrtp 2.1.0.
BUG=webrtc:7856
Review-Url: https://codereview.webrtc.org/2958123002
Cr-Commit-Position: refs/heads/master@{#18805}
The conversion code was wrong because it assumed the 3x3 matrix is a
XYZ-matrix when it really is XYW-matrix. We have to override the matrix
for I420 frames to flip the vertically before rendering.
R=magjed@webrtc.org
Bug: webrtc:7760
Change-Id: I1f08c1a929bf5721706e2a902701100cf7a9c31d
Reviewed-on: https://chromium-review.googlesource.com/541346
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18801}
Adds a timeout to the dequeue input buffer call. This improves stability
because WebRTC quickly queues frames multiple when the call starts. This
might cause the decoder to run out of input buffers. Waiting for
dequeueOutputBuffers call is no longer necessary.
Bug: webrtc:7760
Change-Id: I503ff1cf44042c4d8610077090148d9dfef169f5
Reviewed-on: https://chromium-review.googlesource.com/548357
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18800}
A recent update of Windows 10 blocks IDXGIAdapter::EnumOutputs() in session 0,
so ScreenCapturerWinDirectx::IsSupported() always returns false in session 0. We
should ensure ScreenCapturerWinDirectx can respond correctly in session 0.
Meanwhile, this change looses the requirement of DirectX capturer: it still
works if some of the video adapters do not support DirectX 11 or
IDXGIOutputDuplication. This issue usually happens when handling a virtual video
adapter.
BUG=webrtc:7809
Review-Url: https://codereview.webrtc.org/2937663003
Cr-Commit-Position: refs/heads/master@{#18797}
This change also wires up the rest of the production code in
webrtc/sdk/android to be built when the directory is a dependency.
BUG=webrtc:7613
NOTRY=True
Change-Id: Ideda181970a5a570c3f8148b033e471e926243d1
Reviewed-on: https://chromium-review.googlesource.com/548038
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18791}
