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admin/webrtc_m130

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Adding bit exactness test for Opus decoding in NetEq. 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}

2015-12-22 09:57:41 -08:00

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