webrtc/modules/audio_coding
Jakob Ivarsson 87977dd06e Change buffer level filter to store current level in number of samples.
The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.

Bug: webrtc:10736
Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28368}
2019-06-25 11:21:51 +00:00
..
acm2 Change buffer level filter to store current level in number of samples. 2019-06-25 11:21:51 +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 Change buffer level filter to store current level in number of samples. 2019-06-25 11:21:51 +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