commit 801500cf993582b48836106b40a39d4ed611378c
author Karl Wiberg <kwiberg@webrtc.org> Thu Aug 16 15:01:12 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Aug 22 07:48:55 2018 +0000
tree de44f7937a0e6d04ff53e9b491c208ff3d240bf5
parent 1165949341b6f61c5d728999bfbdaf68fd5c15aa

Audio encoder tests: Create audio encoders the new way

Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.

The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.

I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.

Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}