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webrtc/modules/audio_coding
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Karl Wiberg a1d1a1e976 WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 
Plus tests for 16 kHz.

Bug: webrtc:10631
Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28099}
2019-05-29 10:33:03 +00:00
..
acm2 Audio coding: Don't choke when RTP timestamp rate > sample rate 2019-05-21 03:10:49 +00:00
audio_network_adaptor Stop DCHECK which occurs in ANA BitrateController when overhead is zero. 2019-04-27 00:20:37 +00:00
codecs WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
include Expose new audio stats on the API 2019-05-03 10:10:15 +00:00
neteq Improve NetEq network adaptation in the beginning of the call. 2019-05-23 14:19:30 +00:00
test WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
audio_coding.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
BUILD.gn Encoder side of Multistream Opus. 2019-04-25 15:07:38 +00:00
DEPS
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00