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webrtc_m130/webrtc/video
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Peter Boström 65e7d4cf20 Remove CanCreateAndDestroyManyVideoStreams. 
This test was used to verify that VideoEngine handles were handed back
correctly. This is no longer applicable.

BUG=webrtc:1695
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1425673002 .

Cr-Commit-Position: refs/heads/master@{#10412}
2015-10-26 16:39:43 +00:00
..
BUILD.gn
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
encoded_frame_callback_adapter.cc
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +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
Remove CanCreateAndDestroyManyVideoStreams.
2015-10-26 16:39:43 +00:00
full_stack_plot.py
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
full_stack.cc
Revert of Add screenshare perf tests with lossy links (patchset #1 id:1 of https://codereview.webrtc.org/1409513005/ )
2015-10-19 10:59:13 +00:00
OWNERS
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
rampup_tests.cc
Poll stats more often to get more stable stats in ramp-up tests.
2015-10-05 12:59:50 +00:00
rampup_tests.h
Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc.
2015-09-29 09:26:50 +00:00
receive_statistics_proxy.cc
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
receive_statistics_proxy.h
Add stats for average QP per frame for VP8 (for received video streams):
2015-10-21 06:55:32 +00:00
replay.cc
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
screenshare_loopback.cc
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
send_statistics_proxy_unittest.cc
Remove callback_cs_ in ViEEncoder.
2015-09-22 14:29:00 +00:00
send_statistics_proxy.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
send_statistics_proxy.h
Add histogram for percentage of sent frames that are limited in resolution due to bandwidth:
2015-10-20 06:32:48 +00:00
video_capture_input_unittest.cc
Remove MockVideoCapturer
2015-10-21 16:56:14 +00:00
video_capture_input.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_capture_input.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
video_decoder_unittest.cc
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_decoder.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_encoder_unittest.cc
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_encoder.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_loopback.cc
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
video_quality_test.cc
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
video_quality_test.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
video_receive_stream.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_receive_stream.h
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
video_send_stream_tests.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_send_stream.cc
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
video_send_stream.h
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
webrtc_video.gypi
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00