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webrtc/modules/audio_coding
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Karl Wiberg eb16697259 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate 
A later change will allow them to differ.

Bug: webrtc:10631
Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27991}
2019-05-20 17:33:56 +00:00
..
acm2 Expose new audio stats on the API 2019-05-03 10:10:15 +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 Add DecelerationTargetLevelOffset Field Trial. 2019-05-17 08:08:12 +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
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00