webrtc/modules/audio_coding
Jakob Ivarsson 00a6ab568b Check timestamp difference when choosing to extract multiple packets from the jitter buffer.
This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls.

Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1
Bug: webrtc:10167
Reviewed-on: https://webrtc-review.googlesource.com/c/115419
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26179}
2019-01-09 16:21:11 +00:00
..
acm2 Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +00:00
audio_network_adaptor Use ordered data structure for supported frame lengths in ANA. 2019-01-03 11:56:09 +00:00
codecs Remove CodecInst pt.3 2018-12-18 07:42:21 +00:00
include Remove CodecInst pt.2 2018-12-17 10:33:55 +00:00
neteq Check timestamp difference when choosing to extract multiple packets from the jitter buffer. 2019-01-09 16:21:11 +00:00
test Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +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 Delete NetEq::RegisterExternalDecoder. 2019-01-09 10:38:08 +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