Sergey Silkin 853e247fbb Set full path to input video in EncodeDecode test
Replaced --video_name with --input_path, --input_width, --input_height and --input_framerate_fps.

Example of command line:
video_codec_perf_tests --input_width=1920 --input_height=1080 --input_framerate_fps=30 --input_path="/full/path/sample.yuv"

Also added y4m source support to the codec tester.

Bug: b/42225151, b/337757868
Change-Id: Ic399b3e819a126e097072c27d74a5fcc0dc1fe6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349581
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42226}
2024-05-03 17:46:05 +00:00
2024-05-03 10:39:37 +00:00
2024-05-03 11:36:15 +00:00
2023-10-30 14:56:36 +00:00
2024-04-30 08:47:29 +00:00
2024-04-30 08:47:29 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-04-26 09:56:00 +00:00

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.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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