Henrik Lundin 32f64d2ef9 rtp_encode: Fixing bug related to DTX ...

Bug: webrtc:2692
Change-Id: I7b884b22cab21b9dce77e5599f43431bbc899f5d
Reviewed-on: https://webrtc-review.googlesource.com/26027
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20909}

2017-11-28 12:35:38 +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

rtp_encode: Fixing bug related to DTX

2017-11-28 12:35:38 +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 RTPjitter with a new rtp_jitter

2017-11-24 13:38:59 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

OWNERS

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