webrtc/modules/audio_coding
Minyue Li 3f2eeb8136 Adding test on GetSpanSamples() for NetEq PacketBuffer.
Bug: webrtc:10736
Change-Id: I4448c5c8e1ae8ea5e343415c4fc2c0fd095ca8ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144560
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28481}
2019-07-04 09:23:27 +00:00
..
acm2 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. 2019-07-01 15:56:40 +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 Refactor WebRtcOpus_PacketHasFec. 2019-07-04 07:51:52 +00:00
include Expose new audio stats on the API 2019-05-03 10:10:15 +00:00
neteq Adding test on GetSpanSamples() for NetEq PacketBuffer. 2019-07-04 09:23:27 +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 Improve buffer level estimation with DTX and add CNG time stretching. 2019-07-03 15:12:09 +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