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webrtc_m130/modules/audio_coding
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Karl Wiberg 126f2b37ac AudioEncoderOpus: Add support for 16 kHz input sample rate 
In addition to the 48 kHz that we've always used.

Bug: webrtc:10631
Change-Id: I5e4f6600e39a463d20d3988db098c7e38281f4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138264
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28074}
2019-05-27 13:01:04 +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
AudioEncoderOpus: Add support for 16 kHz input sample rate
2019-05-27 13:01:04 +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 encode sample rate
2019-05-22 22:56:58 +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