Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
8a3b9bf435b7330f472468ed114d9f929f2aab3d
/
test
/
call_test.cc
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
7f57788
Removes trial to enable BBR congestion controller.
by Sebastian Jansson
· 6 years ago
1c931c4
Use VideoSourceInterface for owning test video sources
by Niels Möller
· 6 years ago
8eeccbe
Delete Start and Stop methods from TestVideoCapturer.
by Niels Möller
· 6 years ago
f7f13e0
Add end-to-end test for ColorSpace information
by Johannes Kron
· 6 years ago
de8e6e6
Refactor bitrate configuration in CallTest
by Niels Möller
· 6 years ago
cb7eddb
Add tests for cpu overuse scaling.
by Åsa Persson
· 6 years ago
c2ebe21
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`"
by Jiawei Ou
· 6 years ago
59844ce
Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`."
by Qingsi Wang
· 6 years ago
be14217
Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
by Jiawei Ou
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
75e3647
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
by Artem Titov
· 6 years ago
cbcbc22
Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Niels Möller
· 6 years ago
377b26e
Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Sebastian Jansson
· 6 years ago
efb94d5
Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.""
by Oleh Prypin
· 6 years ago
7961dc2
Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Niels Moller
· 6 years ago
529d0d9
Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
by Niels Möller
· 6 years ago
569397f
Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
by philipel
· 6 years ago
6f68324
Revert "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
by Lu Liu
· 6 years ago
3f4a4fa
Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
by philipel
· 6 years ago
cb7e1d2
Use SdpVideoFormat in VideoReceiveStream::Decoder
by Niels Möller
· 6 years ago
b0588e6
Reland "Move FakeCodec to separate target and behave like real encoder."
by Ilya Nikolaevskiy
· 6 years ago
8d92e8d
Revert "Reland "Move FakeCodec to separate target and behave like real encoder.""
by Ilya Nikolaevskiy
· 6 years ago
f2a8287
Reland "Move FakeCodec to separate target and behave like real encoder."
by Ilya Nikolaevskiy
· 6 years ago
7d13a6e
Revert "Move FakeCodec to separate target and behave like real encoder."
by Ilya Nikolaevskiy
· 6 years ago
223eba5
Move FakeCodec to separate target and behave like real encoder.
by Ilya Nikolaevskiy
· 6 years ago
4e199e9
Mark DirectTransport subclasses ctors as deprecated and switch from them
by Artem Titov
· 6 years ago
46c4e60
Introduce SimulatedNetworkReceiverInterface.
by Artem Titov
· 6 years ago
50eb4c4
Adds BBR field trial to CallTest.
by Sebastian Jansson
· 6 years ago
7258224
Replaces call config create in tests with modify.
by Sebastian Jansson
· 6 years ago
3bd2c79
Moving functionality from VideoQualityTest to CallTest
by Sebastian Jansson
· 6 years ago
f33905d
Makes some CallTest members private.
by Sebastian Jansson
· 6 years ago
8e6602f
Separates send and receive event log in CallTest.
by Sebastian Jansson
· 6 years ago
918f50c
Use absl::make_unique and absl::WrapUnique directly
by Karl Wiberg
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
431abd9
Replace rtc::Optional with absl::optional in test and rtc_tools
by Danil Chapovalov
· 6 years ago
dfce03a
Allows injection of network controller factory into peer connection factory.
by Sebastian Jansson
· 6 years ago
0a8f435
Move VideoEncoderConfig from call/ to api/.
by Niels Möller
· 6 years ago
49fcc10
Merge DegradationPreference enums.
by Taylor Brandstetter
· 6 years ago
4db138e
Reland "Move creating encoder to VideoStreamEncoder."
by Niels Möller
· 7 years ago
0d650b4
Revert "Move creating encoder to VideoStreamEncoder."
by Niels Moller
· 7 years ago
fb82fcc
Move creating encoder to VideoStreamEncoder.
by Niels Möller
· 7 years ago
259a497
Reland "Reland "Move rtp-specific config out of EncoderSettings.""
by Niels Möller
· 7 years ago
6c2c13a
Revert "Reland "Move rtp-specific config out of EncoderSettings.""
by Niels Möller
· 7 years ago
2784a03
Add audio_ prefix to audio-related members of CallTest.
by Niels Möller
· 7 years ago
04dd176
Reland "Move rtp-specific config out of EncoderSettings."
by Niels Möller
· 7 years ago
92be1ca
Revert "Move rtp-specific config out of EncoderSettings."
by Niels Moller
· 7 years ago
bc900cb
Move rtp-specific config out of EncoderSettings.
by Niels Möller
· 7 years ago
465a5d9
Refactor payload types constants in CallTest
by Ilya Nikolaevskiy
· 7 years ago
3faa832
Separate test/fake_audio_device on API and implementation. Step 2.
by Artem Titov
· 7 years ago
03e6ec9
Reland "Add multiplex case to webrtc_perf_tests"
by Emircan Uysaler
· 7 years ago
081136f
Revert "Reland "Add multiplex case to webrtc_perf_tests""
by Taylor Brandstetter
· 7 years ago
7c5bc1c
Reland "Add multiplex case to webrtc_perf_tests"
by Emircan Uysaler
· 7 years ago
5aac372
Revert "Add multiplex case to webrtc_perf_tests"
by Emircan Uysaler
· 7 years ago
d90a7e8
Add multiplex case to webrtc_perf_tests
by Emircan Uysaler
· 7 years ago
6723cdc
Revert "Separate test/fake_audio_device on API and implementation."
by Artem Titov
· 7 years ago
8ea5f9a
Separate test/fake_audio_device on API and implementation.
by Artem Titov
· 7 years ago
9a03dd8
Removed new calls on RtpTransportControllerSend.
by Sebastian Jansson
· 7 years ago
97f61ea
Moved bitrate configuration to rtp controller
by Sebastian Jansson
· 7 years ago
3f6804d
Optional: Use nullopt and implicit construction in /test
by Oskar Sundbom
· 7 years ago
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
62337e5
Use AudioProcessingBuilder everywhere AudioProcessing is created.
by Ivo Creusen
· 7 years ago
255d1cd
Implement dual stream full stack test and loopback tool
by Ilya Nikolaevskiy
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
d319534
Move ADM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
3102734
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."
by Rasmus Brandt
· 7 years ago
2666cf7
Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
by Rasmus Brandt
· 7 years ago
2c30120
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
by brandtr
· 7 years ago
2cefac6
Add full stack tests for MediaCodec encoder.
by brandtr
· 7 years ago
7cd28b9
Set protected_by_flexfec flag properly in tests.
by brandtr
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/test/call_test.cc]
73276ad
- Removes voe_conference_test.
by Fredrik Solenberg
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
413ee9a
Use SingleThreadedTaskQueue in DirectTransport
by eladalon
· 7 years ago
2bf9e73
Delete unneeded Start and Stop methods on FlexfecReceiveStream.
by Niels Möller
· 7 years ago
db2a9fc
Wire up RTP keep-alive in ortc api.
by sprang
· 7 years ago
445f1a1
nit: Order CallTest's methods in the .cc according to their order in the .h file.
by eladalon
· 7 years ago
c0d481a
Protected streams report RTP messages directly to the FlexFec streams
by eladalon
· 7 years ago
863f03b
Fix video_replay tool to respect RTX stream and fix default parameters.
by ilnik
· 7 years ago
d2702ef
Fix flaky test VideoSendStreamTest.SendsKeepAlive
by sprang
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 7 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 8 years ago
00d802b
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
by ilnik
· 8 years ago
27c46e2
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
by ilnik
· 8 years ago
774f6b4
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
29dbb19
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
by ilnik
· 8 years ago
4fa0c4f
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
by ilnik
· 8 years ago
5721866
Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
by ilnik
· 8 years ago
64e739a
Add content type information to Encoded Images and add corresponding RTP extension header.
by ilnik
· 8 years ago
20c84cc
Making FakeNetworkPipe demux audio and video packets.
by minyue
· 8 years ago
4fb651d
Event log cleanup in tests.
by philipel
· 8 years ago
d8ce1e1
Move SelectMediaType from RampUpTester to BaseTest.
by nisse
· 8 years ago
c5d62e2
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
by sprang
· 8 years ago
f9ed235
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
by lliuu
· 8 years ago
3ea3c77
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
by sprang
· 8 years ago
Next »