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webrtc_m130/modules/audio_coding
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Jakob Ivarsson 9ce451a03f End NetEq simulation if there are no more packets to decode. 
Bug: b/133217334
Change-Id: Ibd696011f390ef60a6ac44e603ab4380ae5e759a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138060
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28026}
2019-05-22 15:21:44 +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: Don't mix up sample rate and RTP timestamp rate
2019-05-20 17:33:56 +00:00
include
Expose new audio stats on the API
2019-05-03 10:10:15 +00:00
neteq
End NetEq simulation if there are no more packets to decode.
2019-05-22 15:21:44 +00:00
test
Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData
2019-04-26 12:58:14 +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
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OWNERS
Make ivoc owner of audio_coding.
2018-10-15 15:08:28 +00:00