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webrtc/modules/audio_coding
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Karl Wiberg 7eb0a5e210 AudioDecoderOpus: Add support for 16 kHz output sample rate 
In addition to the 48 kHz that we've always used.

Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
2019-05-29 12:42:38 +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 AudioDecoderOpus: Add support for 16 kHz output sample rate 2019-05-29 12:42:38 +00:00
include Expose new audio stats on the API 2019-05-03 10:10:15 +00:00
neteq AudioDecoderOpus: Add support for 16 kHz output sample rate 2019-05-29 12:42:38 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00