2 commits

Author	SHA1	Message	Date
kjellander	c3a0983d4b	Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2 ... Includes updates to tests for Opus v.1.1.2, reveiwed in https://codereview.webrtc.org/1629413002/ Change log: a8e5140..c6076f2 Full diff: a8e5140..c6076f2 Changed dependencies: * src/third_party/catapult: 471db30..d4d48e6 * src/third_party/opus/src: cae6961..655cc54 DEPS diff: a8e5140..c6076f2/DEPS No update to Clang. BUG=chromium:580524 TBR= Review URL: https://codereview.webrtc.org/1657343002 Cr-Commit-Position: refs/heads/master@{#11464}	2016-02-02 21:18:42 +00:00
minyue	93c08b7438	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 17:57:47 +00:00