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

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