93c08b7438
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq.
The new RTP file is generated by the following steps:
1. Encode a clean RTP file with Opus
RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1
2. Adding jitter to the clean RTP file
RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp
(Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.)
BUG=webrtc:3987
TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output.
Review URL: https://codereview.webrtc.org/1515113002
Cr-Commit-Position: refs/heads/master@{#11113}
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc