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webrtc_m130/webrtc/video
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Peter Boström 5cb9ce4c74 Remove ViECodec usage in VideoSendStream. 
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also inlining the necessary parts of SetSendCodec
done previously in ViECodecImpl.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46129004

Cr-Commit-Position: refs/heads/master@{#9136}
2015-05-05 13:16:40 +00:00
..
audio_receive_stream.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
bitrate_estimator_tests.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
BUILD.gn
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
call_perf_tests.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
call.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
encoded_frame_callback_adapter.cc
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
2015-03-23 13:13:15 +00:00
encoded_frame_callback_adapter.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
end_to_end_tests.cc
Use padding to achieve bitrate probing if the initial key frame has too few packets.
2015-05-05 08:21:32 +00:00
full_stack.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
loopback.cc
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
2015-04-21 12:25:42 +00:00
loopback.h
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
OWNERS
Adding pbos as video/ owner and removing persons never working with this folder.
2014-07-04 08:50:14 +00:00
rampup_tests.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
rampup_tests.h
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
receive_statistics_proxy.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
receive_statistics_proxy.h
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
replay.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
screenshare_loopback.cc
Fix dangling pointer in screenshare_loopback
2015-04-28 12:00:29 +00:00
send_statistics_proxy_unittest.cc
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
send_statistics_proxy.cc
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
send_statistics_proxy.h
Use rtc::CriticalSection in webrtc/video/.
2015-05-01 14:25:53 +00:00
transport_adapter.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
transport_adapter.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_loopback.cc
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
video_receive_stream.cc
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
video_receive_stream.h
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
video_send_stream_tests.cc
Use padding to achieve bitrate probing if the initial key frame has too few packets.
2015-05-05 08:21:32 +00:00
video_send_stream.cc
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
video_send_stream.h
Remove ViECodec usage in VideoSendStream.
2015-05-05 13:16:40 +00:00
webrtc_video.gypi
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00