commit | 5d6a06c1d29a2061bcf4b321ffceab477a404d51 | [log] [tgz] |
---|---|---|
author | ivica <ivica@webrtc.org> | Thu Sep 17 12:30:24 2015 |
committer | Commit bot <commit-bot@chromium.org> | Thu Sep 17 12:30:30 2015 |
tree | 7ab0670f5686db87e1f5508f358bc2ba7e446172 | |
parent | f2bfc2b8ef3d774658b9ce3dcd6757f932d071fb [diff] |
Refactoring full stack and loopback tests Refactoring full stack, video and screenshare tests to use the same code basis for parametrization and initialization. This patch is done on top of recently commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer in full stack, except moving it to video_quality_test.cc. Also, full_stack_samples.cc (build target) was removed and replaced with -output_filename and -duration cmdline arguments in video_loopback and screenshare_loopback. The important things to review: - video_quality_test.h Is the structure of Params good? (examples of usage can be found in full_stack.cc, video_loopback.cc and screenshare_loopback.cc) - video_quality_test.cc Is the initialization correct? The case for using Analyzer and using local renderer are different, can they be further merged? - webrtc_tests.gypi Reproducing the different bitrate settings the full stack and loopback tests had was a little bit tricky. To support both simultaneously, I added BitrateConfig to the Params struct, as well as separate start_bitrate and target_bitrate flags for loopback tests. Note: Side-by-side diff for video_quality_test.cc compares that file directly with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible. Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold args to loopback tests. This was removed here. Support for streams and SVC will be added in a CL following this one. Review URL: https://codereview.webrtc.org/1308403003 Cr-Commit-Position: refs/heads/master@{#9969}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.