The ConversationalSpeechTest.* unit tests are now part of modules_unittests.
The rtc_test target has been replaced with an rtc_source_set one.
The latter is included as dependency in audio_processing_unittests.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2769863005
Cr-Commit-Position: refs/heads/master@{#17360}
The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout.
The test cases are meant to try different network and audio configs (more are planned in the future).
The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard.
BUG=webrtc:7229
NOTRY=True
Review-Url: https://codereview.webrtc.org/2694203002
Cr-Commit-Position: refs/heads/master@{#17356}
constexpr function should be preferred than a macro. So this change replaces
FOURCC() macro with a constexpr uint32_t FourCC() function.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2771573002
Cr-Commit-Position: refs/heads/master@{#17351}
Moves towards separating which layers may be referenced instead of
referencing libvpx flags directly. This will make strategies easier to
extract and usable from hardware encoders (RTCVideoEncoder, for
instance).
BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org, marpan@webrtc.org. sprang@webrtc.org
Review-Url: https://codereview.webrtc.org/2747123005
Cr-Commit-Position: refs/heads/master@{#17349}
This change adds a DesktopCapturerId namespace, and attaches an int to each
DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
this field to differentiate themselves.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Original-Commit-Position: refs/heads/master@{#17329}
Committed: 41e3d9ff3b
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Commit-Position: refs/heads/master@{#17347}
The conversational_speech::Timing class models a list of turns.
Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds.
The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2750353002
Cr-Commit-Position: refs/heads/master@{#17346}
This test was not built by default so it fails when it's now
enabled as a sanity check on the "more configs" bots.
BUG=webrtc:7228
TBR=nisse@webrtc.org
Review-Url: https://codereview.webrtc.org/2765843004 .
Cr-Commit-Position: refs/heads/master@{#17343}
This cl change the drawing to use memset per row in each square instead of setting each pixel individually.
BUG=webrtc:7380
Review-Url: https://codereview.webrtc.org/2764753003
Cr-Commit-Position: refs/heads/master@{#17339}
Reason for revert:
Problem was the rename of the include file. Intend to keep the old name for now, and then reland.
Original issue's description:
> Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ )
>
> Reason for revert:
> This is breaking downstream build.
>
> Original issue's description:
> > Delete class MockCongestionController.
> >
> > It became unused in cl https://codereview.webrtc.org/2516983004.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2762023004
> > Cr-Commit-Position: refs/heads/master@{#17325}
> > Committed: d19bcb7116
>
> TBR=stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2762133003
> Cr-Commit-Position: refs/heads/master@{#17330}
> Committed: e27f1e764eTBR=stefan@webrtc.org,skvlad@webrtc.org
BUG=None
Review-Url: https://codereview.webrtc.org/2766133002
Cr-Commit-Position: refs/heads/master@{#17338}
Reason for revert:
I suspect that this CL breaks Chromium WebRTC FYI bots. (Thanks kjellander@ for spotting.) The added dep in BUILD.gn would be the problem.
Example:
https://luci-logdog.appspot.com/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F15058%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
FAILED: newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
/b/c/goma_client/gomacc ../../native_client/toolchain/linux_x86/pnacl_newlib/bin/pnacl-clang++ -MMD -MF newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o.d -DNACL_TC_REV=5dfe030a71ca66e72c5719ef5034c2ed24706c43 -DV8_DEPRECATION_WARNINGS -DUSE_OPENSSL_CERTS=1 -DNO_TCMALLOC -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL -DCHROMIUM_BUILD -DENABLE_MEDIA_ROUTER=1 -DFIELDTRIAL_TESTING_ENABLED -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -D_FORTIFY_SOURCE=2 -DNDEBUG -DNVALGRIND -DWEBRTC_RESTRICT_LOGGING -DEXPAT_RELATIVE_PATH -DHAVE_SCTP -DENABLE_EXTERNAL_AUTH -DHAVE_WEBRTC_VIDEO -DHAVE_WEBRTC_VOICE -DLOGGING_INSIDE_WEBRTC -DUSE_WEBRTC_DEV_BRANCH -DFEATURE_ENABLE_VOICEMAIL -DEXPAT_RELATIVE_PATH -DGTEST_RELATIVE_PATH -DNO_MAIN_THREAD_WRAPPING -DNO_SOUND_SYSTEM -DWEBRTC_CHROMIUM_BUILD -DWEBRTC_POSIX -I../.. -Inewlib_pnacl/gen -I../../third_party/webrtc_overrides -I../../third_party -fno-strict-aliasing -Wno-builtin-macro-redefined -D__DATE__= -D__TIME__= -D__TIMESTAMP__= -fcolor-diagnostics -Wall -Werror -Wextra -Wno-missing-field-initializers -Wno-unused-parameter -Wno-c++11-narrowing -Wno-covered-switch-default -Wno-deprecated-register -Wno-unneeded-internal-declaration -Wno-inconsistent-missing-override -O2 -fno-ident -fdata-sections -ffunction-sections -g0 -fvisibility=hidden -fvisibility-inlines-hidden -std=gnu++11 -fno-rtti -fno-exceptions -c ../../third_party/webrtc/api/mediaconstraintsinterface.cc -o newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.cc:11:
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.h:27:
In file included from ../../third_party/webrtc/api/peerconnectioninterface.h:77:
In file included from ../../third_party/webrtc/api/dtmfsenderinterface.h:16:
In file included from ../../third_party/webrtc/api/mediastreaminterface.h:33:
In file included from ../../third_party/webrtc/media/base/mediachannel.h:28:
../../third_party/webrtc/base/socket.h:18:10: fatal error: 'sys/socket.h' file not found
#include <sys/socket.h>
^
1 error generated.
Original issue's description:
> Add DesktopCapturerId and attach it to DesktopFrame
>
> This change adds a DesktopCapturerId namespace, and attaches an int to each
> DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
> this field to differentiate themselves.
>
> BUG=679523, 650926
>
> Review-Url: https://codereview.webrtc.org/2759493002
> Cr-Commit-Position: refs/heads/master@{#17329}
> Committed: 41e3d9ff3bTBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2767003002
Cr-Commit-Position: refs/heads/master@{#17336}
This was causing the QualityScaler to be reconstructed each time
the resolution changes and thus the fast_rampup logic was not working
as intended. We now properly change the checking period to 5 seconds
after a downscale.
BUG=b/36457883
Review-Url: https://codereview.webrtc.org/2766513003
Cr-Commit-Position: refs/heads/master@{#17335}
This change adds a DesktopCapturerId namespace, and attaches an int to each
DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
this field to differentiate themselves.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Commit-Position: refs/heads/master@{#17329}
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
Extract the remote addresses from SDP c= line on both session level and
media level. The media level address will overwrite the session level one if
exists.
WebRTC is not using c= and this is used for new SDP parsing API.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2742903002
Cr-Commit-Position: refs/heads/master@{#17326}
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.
Rest of the CongestionController moved to a new class
SendSideCongestionController.
To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.
With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
Add tests for inital probing and mid-call probing by reconfiguring max bitrate.
BUG=none
Review-Url: https://codereview.webrtc.org/2760623002
Cr-Commit-Position: refs/heads/master@{#17316}
The only thing that was holding us back was the indeterministic teardown of voe::Channel(), but it turned out that fixing it wasn't that hard :)
BUG=webrtc:4508
Review-Url: https://codereview.webrtc.org/2755273004
Cr-Commit-Position: refs/heads/master@{#17315}
It depends on RTCP RPSI and SLI messages, which are being deleted.
TBR=stefan@webrtc.org # TODO comments added to common_types.h
BUG=webrtc:7338
Review-Url: https://codereview.webrtc.org/2753783002
Cr-Commit-Position: refs/heads/master@{#17314}
- Reduced flakyness by switching to a packetization format that has
PictureID.
- Removed the explicit send-side BWE enabling from ULPFEC and FlexFEC
tests, since that is now on by default.
BUG=webrtc:7047
Review-Url: https://codereview.webrtc.org/2753253002
Cr-Commit-Position: refs/heads/master@{#17310}
This is one step towards separation of send-side and receive-side
processing.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2740163002
Cr-Commit-Position: refs/heads/master@{#17306}
This DCHECK is for the 'new and improved' way of setting thread priority.
What could happen is that code that's migrating over to the new method
might still have a lingering SetPriority call, that could incorrectly bind the
'spawned_thread_checker_' to the construction thread.
BUG=none
Review-Url: https://codereview.webrtc.org/2753423002
Cr-Commit-Position: refs/heads/master@{#17305}
This is a problem if a data channel is re-opened or a new data channel
occupies the same space in memory as a previously closed data channel.
Unittest updated to cover this (failed before fix, now passes).
BUG=webrtc:7181
Review-Url: https://codereview.webrtc.org/2746393003
Cr-Commit-Position: refs/heads/master@{#17304}
The results of the echo detector complexity tests are currently notoriously spiky and unreliable. The following improvements are made in this CL:
- Significantly longer warmup time before starting the test
- More iterations and larger batches
- Different number of iterations for slow and fast tests
- Use the echo likelihood in the test so it cannot get optimized out
BUG=webrtc:7353
Review-Url: https://codereview.webrtc.org/2750413002
Cr-Commit-Position: refs/heads/master@{#17303}
Test for the conversational_speech::Config class and renaming.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2749573002
Cr-Commit-Position: refs/heads/master@{#17301}
generation."
This CL removes the Python script and adds its C++ porting.
The former was in its early stage and it has permanently been removed.
This is a reland of https://codereview.webrtc.org/2740063004/ which
was reverted. Now the build errors are fixed.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2752793002
Cr-Commit-Position: refs/heads/master@{#17300}
SrtpTransportInterface methods take cricket::CryptoParams, so this
should be enough for now.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2753343002
Cr-Commit-Position: refs/heads/master@{#17299}
CreateRawScreenCapturer() and CreateRawWindowCapturer() in
DesktopCapturer are allowed to return nullptr when capturer cannot be
initialized for some reason. CreateWindowCapturer() and
CreateScreenCapturer() in DesktopCapturer were not handling this case
correctly, which may lead to crash.
BUG=chromium:702745
Review-Url: https://codereview.webrtc.org/2754403002
Cr-Commit-Position: refs/heads/master@{#17298}
Reason for revert:
Reverting this as it had no effect.
Original issue's description:
> Don't set the priority of the decoder to 'high' on Android.
> Doing so competes with the actual decoding that happens on a different thread.
>
> BUG=695438
>
> Review-Url: https://codereview.webrtc.org/2745813003
> Cr-Commit-Position: refs/heads/master@{#17184}
> Committed: ca37cf6691TBR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=695438
Review-Url: https://codereview.webrtc.org/2757733005
Cr-Commit-Position: refs/heads/master@{#17297}
