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webrtc_m130/webrtc/video
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sergeyu 80ed35e21c Implement periodic bandwidth probing in application-limited region. 
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.

BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
2016-11-28 21:11:24 +00:00
..
BUILD.gn
Pass selected cricket::VideoCodec down to internal H264 encoder
2016-11-28 15:20:26 +00:00
call_stats_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
call_stats.cc
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
call_stats.h
Replace scoped_ptr with unique_ptr in webrtc/video/
2016-03-01 19:52:39 +00:00
DEPS
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
encoder_rtcp_feedback_unittest.cc
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
encoder_rtcp_feedback.cc
Remove OnLocalSsrcChanged and rename EncoderStateFeedback.
2016-10-06 15:35:19 +00:00
encoder_rtcp_feedback.h
Remove OnLocalSsrcChanged and rename EncoderStateFeedback.
2016-10-06 15:35:19 +00:00
end_to_end_tests.cc
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +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
Reduce full stack test time to 45 secs and add H264 and FlexFEC.
2016-11-17 07:57:00 +00:00
overuse_frame_detector_unittest.cc
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
overuse_frame_detector.cc
Move smoothing filter to common audio and exp_filter to base/analytics.
2016-11-24 19:01:14 +00:00
overuse_frame_detector.h
Move smoothing filter to common audio and exp_filter to base/analytics.
2016-11-24 19:01:14 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
payload_router_unittest.cc
Fix PayloadRouter::OnEncodedImage() to handle errors properly.
2016-11-18 00:16:22 +00:00
payload_router.cc
Fix PayloadRouter::OnEncodedImage() to handle errors properly.
2016-11-18 00:16:22 +00:00
payload_router.h
Let ViEEncoder tell VideoSendStream about reconfigurations.
2016-09-23 13:52:20 +00:00
receive_statistics_proxy_unittest.cc
Add unit tests to ReceiveStatisticsProxy class.
2016-11-03 13:48:36 +00:00
receive_statistics_proxy.cc
Add ToString method to AggregatedStats and log stats at the end of a call.
2016-11-15 16:20:54 +00:00
receive_statistics_proxy.h
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
replay.cc
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
report_block_stats_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
report_block_stats.cc
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
report_block_stats.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
rtp_stream_receiver.cc
Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
2016-11-25 14:40:30 +00:00
rtp_stream_receiver.h
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
rtp_streams_synchronizer.cc
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
rtp_streams_synchronizer.h
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
screenshare_loopback.cc
Configure FlexFEC in VideoQualityTest.
2016-11-17 06:47:36 +00:00
send_delay_stats_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
send_delay_stats.cc
Add ToString method to AggregatedStats and log stats at the end of a call.
2016-11-15 16:20:54 +00:00
send_delay_stats.h
Start integrating StatsCounter class.
2016-06-09 07:09:27 +00:00
send_statistics_proxy_unittest.cc
Sent bitrate stats are incorrect if FlexFEC is configured:
2016-11-25 11:52:55 +00:00
send_statistics_proxy.cc
Sent bitrate stats are incorrect if FlexFEC is configured:
2016-11-25 11:52:55 +00:00
send_statistics_proxy.h
Move smoothing filter to common audio and exp_filter to base/analytics.
2016-11-24 19:01:14 +00:00
stats_counter_unittest.cc
Add ability to handle data from multiple streams in RateAccCounter.
2016-11-01 07:21:41 +00:00
stats_counter.cc
Add ToString method to AggregatedStats and log stats at the end of a call.
2016-11-15 16:20:54 +00:00
stats_counter.h
Add ToString method to AggregatedStats and log stats at the end of a call.
2016-11-15 16:20:54 +00:00
stream_synchronization_unittest.cc
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
stream_synchronization.cc
Move RTP for synchroninzation and rename classes, files and variables.
2016-08-05 13:28:50 +00:00
stream_synchronization.h
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
transport_adapter.cc
Moved transport_adapter.h/.cc from call/ to video/ dir to remove circular dependency
2016-11-03 11:21:47 +00:00
transport_adapter.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_decoder.cc
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_loopback.cc
Integrate FlexFEC in video_loopback.
2016-11-17 08:21:19 +00:00
video_quality_test.cc
Pass selected cricket::VideoCodec down to internal H264 encoder
2016-11-28 15:20:26 +00:00
video_quality_test.h
Configure FlexFEC in VideoQualityTest.
2016-11-17 06:47:36 +00:00
video_receive_stream.cc
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
video_receive_stream.h
New jitter buffer experiment.
2016-11-15 08:58:06 +00:00
video_send_stream_tests.cc
Try to deflake VideoSendStream tests with FlexFEC.
2016-11-23 12:42:31 +00:00
video_send_stream.cc
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
video_send_stream.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
video_stream_decoder.cc
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
video_stream_decoder.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
vie_encoder_unittest.cc
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
2016-11-25 12:37:06 +00:00
vie_encoder.cc
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
2016-11-25 12:37:06 +00:00
vie_encoder.h
Turn off error resilience for vp8 for no temporal layers if nack is enabled.
2016-11-25 12:37:06 +00:00
vie_remb_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
vie_remb.cc
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
vie_remb.h
Replace scoped_ptr with unique_ptr in webrtc/video/
2016-03-01 19:52:39 +00:00