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webrtc_m130/webrtc/video
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nisse 5206667dad Delete unused method PayloadRouter::MaxPayloadLength. 
Documentation was also unclear, it seems it returned the RTP packet
size including RTP headers.

BUG=None.

Review-Url: https://codereview.webrtc.org/2588343002
Cr-Commit-Position: refs/heads/master@{#15707}
2016-12-20 11:12:04 +00:00
..
BUILD.gn
Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
2016-12-19 13:47:28 +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
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
full_stack_tests_plot.py
Rename full_stack.cc to full_stack_tests.cc.
2016-12-01 08:20:24 +00:00
full_stack_tests.cc
Add full stack tests:
2016-12-16 14:54:01 +00:00
overuse_frame_detector_unittest.cc
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
overuse_frame_detector.cc
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
overuse_frame_detector.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
OWNERS
Adding Åsa and Erik as video owners.
2016-12-16 10:12:31 +00:00
payload_router_unittest.cc
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +00:00
payload_router.cc
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +00:00
payload_router.h
Delete unused method PayloadRouter::MaxPayloadLength.
2016-12-20 11:12:04 +00:00
quality_threshold_unittest.cc
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
quality_threshold.cc
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
quality_threshold.h
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
receive_statistics_proxy_unittest.cc
Update video histograms that do not have a minimum lifetime limit before being recorded.
2016-11-30 09:42:32 +00:00
receive_statistics_proxy.cc
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
receive_statistics_proxy.h
Logging basic bad call detection
2016-12-13 10:46:06 +00:00
replay.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
…
report_block_stats.h
…
rtp_stream_receiver.cc
Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
2016-12-13 10:23:43 +00:00
rtp_stream_receiver.h
Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
2016-12-13 10:23:43 +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
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
send_statistics_proxy.cc
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +00:00
send_statistics_proxy.h
Move histogram for number of pause events to per stream:
2016-12-19 14:50:53 +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
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +00:00
stats_counter.h
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +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_loopback.cc
Integrate FlexFEC in video_loopback.
2016-11-17 08:21:19 +00:00
video_quality_test.cc
Fixing loopback video test by reconfiguring the encoder to correct size.
2016-12-15 15:24:38 +00:00
video_quality_test.h
Configure FlexFEC in VideoQualityTest.
2016-11-17 06:47:36 +00:00
video_receive_stream.cc
Create VideoReceiver with external VCMTiming object.
2016-12-15 15:11:01 +00:00
video_receive_stream.h
Create VideoReceiver with external VCMTiming object.
2016-12-15 15:11:01 +00:00
video_send_stream_tests.cc
Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
2016-12-19 17:36:33 +00:00
video_send_stream.cc
Simplify an always true condition.
2016-12-08 14:24:35 +00:00
video_send_stream.h
Move ownership of PacketRouter from CongestionController to Call.
2016-11-30 11:35:28 +00:00
video_stream_decoder.cc
Delete all of the video_processing module but the denoiser code.
2016-11-30 15:44:59 +00:00
video_stream_decoder.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
vie_encoder_unittest.cc
Reland of Bump up scaling limit for MediaCodec. (patchset #1 id:1 of https://codereview.webrtc.org/2562963002/ )
2016-12-09 11:54:08 +00:00
vie_encoder.cc
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +00:00
vie_encoder.h
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +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