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webrtc/modules/audio_coding
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Henrik Lundin 1391bd472c Replacing the legacy tool RTPencode with a new rtp_encode 
This new tool provides the same functionality as the legacy tool, but it
is implemented using AudioCodingModule and AudioEncoder APIs instead of
the naked codecs.

Bug: webrtc:2692
Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc
Reviewed-on: https://webrtc-review.googlesource.com/24861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:05:48 +00:00
..
acm2 Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
audio_network_adaptor Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
codecs Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
include Remove AudioCodingModule::IncomingPayload 2017-09-29 14:23:27 +00:00
neteq Replacing the legacy tool RTPencode with a new rtp_encode 2017-11-24 09:05:48 +00:00
test Avoid flagging Opus DTX frames as speech. 2017-11-20 14:53:40 +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 Replacing the legacy tool RTPencode with a new rtp_encode 2017-11-24 09:05:48 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00