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webrtc_m130/modules/audio_coding
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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
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OWNERS
Make ivoc owner of audio_coding.
2018-10-15 15:08:28 +00:00