admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/call
History
solenberg c7a8b08a7c Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. 
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1397123003

Cr-Commit-Position: refs/heads/master@{#10307}
2015-10-16 21:35:11 +00:00
..
bitrate_estimator_tests.cc
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
BUILD.gn
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
call_perf_tests.cc
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
call_unittest.cc
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
call.cc
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
OWNERS
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
packet_injection_tests.cc
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
rtc_event_log2rtp_dump.cc
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
rtc_event_log_unittest.cc
Old config events are no longer removed from RtcEventLog.
2015-10-16 15:51:15 +00:00
rtc_event_log.cc
Old config events are no longer removed from RtcEventLog.
2015-10-16 15:51:15 +00:00
rtc_event_log.h
Old config events are no longer removed from RtcEventLog.
2015-10-16 15:51:15 +00:00
rtc_event_log.proto
Update to the RtcEventLog protobuf to remove the DebugEvent message.
2015-10-08 16:07:53 +00:00
transport_adapter.cc
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
transport_adapter.h
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
webrtc_call.gypi
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00