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Danil Chapovalov a715f28968 Fix handling invalid empty red packets 
Bug: chromium:856823
Change-Id: I3e64697cd99c6ca67e1102e18ec03965f67d4b9c
Reviewed-on: https://webrtc-review.googlesource.com/88227
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23946}
2018-07-12 07:56:55 +00:00
..
end_to_end_tests
Removes unnecessary destructor checks in tests.
2018-07-10 11:29:45 +00:00
test
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Moved PayloadRouter to call/.
2018-07-11 15:38:39 +00:00
call_stats_unittest.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.cc
Reduce locking for CallStats (preparation for TaskQueue).
2018-03-28 13:24:07 +00:00
call_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
encoder_rtcp_feedback_unittest.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
encoder_rtcp_feedback.cc
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
encoder_rtcp_feedback.h
Adds unit tests for VideoSendStreamImpl.
2018-04-20 13:12:13 +00:00
full_stack_tests_plot.py
…
full_stack_tests.cc
Fix incorrect screenshare bitrate configs in test cases
2018-07-11 15:14:19 +00:00
overuse_frame_detector_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
overuse_frame_detector.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
overuse_frame_detector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
…
picture_id_tests.cc
Removes unnecessary destructor checks in tests.
2018-07-10 11:29:45 +00:00
quality_scaling_tests.cc
Add tests for quality scaling.
2018-05-04 07:51:04 +00:00
quality_threshold_unittest.cc
…
quality_threshold.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
quality_threshold.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
receive_statistics_proxy.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
receive_statistics_proxy.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
replay.cc
Delete pre_decode_callback.
2018-06-20 07:04:09 +00:00
report_block_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
report_block_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_streams_synchronizer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_streams_synchronizer.h
Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
2018-02-07 10:07:28 +00:00
rtp_video_stream_receiver_unittest.cc
Fix handling invalid empty red packets
2018-07-12 07:56:55 +00:00
rtp_video_stream_receiver.cc
Fix handling invalid empty red packets
2018-07-12 07:56:55 +00:00
rtp_video_stream_receiver.h
Remove StreamStatistician::IsPacketInOrder
2018-06-28 08:44:40 +00:00
screenshare_loopback.cc
Fix incorrect screenshare bitrate configs in test cases
2018-07-11 15:14:19 +00:00
send_delay_stats_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_delay_stats.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
send_delay_stats.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
send_statistics_proxy_unittest.cc
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
send_statistics_proxy.cc
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
send_statistics_proxy.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stats_counter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stats_counter.cc
…
stats_counter.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
stream_synchronization_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
stream_synchronization.h
Adding NOLINT for typedefs.h and common_types.h
2017-09-15 13:03:51 +00:00
sv_loopback.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
transport_adapter.cc
Replace Atomic32 with std::atomic in video/
2017-10-24 23:40:29 +00:00
transport_adapter.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_analyzer.cc
Moves VideoAnalyzer class to a separate file.
2018-07-10 11:32:45 +00:00
video_analyzer.h
Moves VideoAnalyzer class to a separate file.
2018-07-10 11:32:45 +00:00
video_loopback.cc
Fix incorrect screenshare bitrate configs in test cases
2018-07-11 15:14:19 +00:00
video_quality_observer.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_observer.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_quality_test.cc
Revert "Add Profile 2 configuration to VP9 Encoder and Decoder"
2018-07-11 06:04:49 +00:00
video_quality_test.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
video_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_receive_stream.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_receive_stream.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_send_stream_impl_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_send_stream_impl.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_send_stream_impl.h
Moved PayloadRouter to call/.
2018-07-11 15:38:39 +00:00
video_send_stream_tests.cc
Stop using VideoCodec.targetBitrate for vp8 screenshare config
2018-07-06 15:13:18 +00:00
video_send_stream.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_send_stream.h
Moved PayloadRouter to call/.
2018-07-11 15:38:39 +00:00
video_stream_decoder_impl.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_stream_decoder_impl.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_stream_decoder.cc
Moved PayloadRouter to call/.
2018-07-11 15:38:39 +00:00
video_stream_decoder.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_stream_encoder_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
video_stream_encoder.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
video_stream_encoder.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00