Logo
Explore Help
Register Sign In
admin/webrtc_m130
1
0
Fork 0
You've already forked webrtc_m130
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/audio
History
michaelt 79e05888e8 Set actual transport overhead in rtp_rtcp
BUG=webrtc:6557

Review-Url: https://codereview.webrtc.org/2437503004
Cr-Commit-Position: refs/heads/master@{#14968}
2016-11-08 10:50:16 +00:00
..
audio_receive_stream_unittest.cc
Clean up abs-send-time for audio.
2016-11-01 10:17:18 +00:00
audio_receive_stream.cc
Fix crash when registering abs-send-time to AudioSend/ReceiveStream.
2016-11-02 10:10:12 +00:00
audio_receive_stream.h
Add a NeededFrequency() method to the AudioMixer::Source interface.
2016-10-31 10:26:48 +00:00
audio_send_stream_unittest.cc
Fixing config for Audio BWE.
2016-11-07 17:29:27 +00:00
audio_send_stream.cc
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
audio_send_stream.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
audio_state_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
audio_state.cc
…
audio_state.h
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
BUILD.gn
Made AudioReceiveStream a mixer participant.
2016-10-20 13:32:47 +00:00
conversion.h
…
DEPS
Fix BWE simulations so that it uses the delay based BWE.
2016-10-25 14:04:44 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
scoped_voe_interface.h
…
webrtc_audio.gypi
Made AudioReceiveStream a mixer participant.
2016-10-20 13:32:47 +00:00
Powered by Gitea Version: 1.23.5 Page: 2647ms Template: 114ms
English
Bahasa Indonesia Deutsch English Español Français Gaeilge Italiano Latviešu Magyar nyelv Nederlands Polski Português de Portugal Português do Brasil Suomi Svenska Türkçe Čeština Ελληνικά Български Русский Українська فارسی മലയാളം 日本語 简体中文 繁體中文(台灣) 繁體中文(香港) 한국어
Licenses API