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webrtc_m130/modules/audio_coding
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Jakob Ivarsson 1b4254ada5 Check current buffer time span instead of number of samples in postpone decoding after expand. 
Bug: webrtc:10392
Change-Id: I2ad4d8c7a3f87cab32e2ea097b2e05aa179e0bc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126761
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27080}
2019-03-12 16:25:24 +00:00
..
acm2
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
audio_network_adaptor
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
codecs
Fix -Wextra-semi warnings.
2019-02-25 09:22:51 +00:00
include
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +00:00
neteq
Check current buffer time span instead of number of samples in postpone decoding after expand.
2019-03-12 16:25:24 +00:00
test
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
2019-03-07 10:12:57 +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
Revert "NetEQ RTP Play: Optionally write output audio file"
2019-03-12 14:28:23 +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