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webrtc_m130/webrtc/call
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Danil Chapovalov bcdad0f3a8 Generate random rtp packets with RtpPacketToSend instead of RtpSender 
in rtc event log unittest

R=stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2228493002 .

Cr-Commit-Position: refs/heads/master@{#13705}
2016-08-10 11:23:38 +00:00
..
mock
Refactored CL for moving the output to a separate thread.
2016-04-22 19:40:46 +00:00
bitrate_allocator_unittest.cc
Auto pause video streams based on encoder target bitrate.
2016-07-01 11:04:10 +00:00
bitrate_allocator.cc
Add more logging about the bwe state and VideoSendStream state.
2016-07-07 07:36:40 +00:00
bitrate_allocator.h
Auto pause video streams based on encoder target bitrate.
2016-07-01 11:04:10 +00:00
bitrate_estimator_tests.cc
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
BUILD.gn
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call_perf_tests.cc
Changes synchronization offset perfomance tracking
2016-06-16 07:58:58 +00:00
call_unittest.cc
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
call.cc
Only update codec type histogram if lifetime is long enough (10 sec).
2016-07-27 07:39:17 +00:00
DEPS
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
OWNERS
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
packet_injection_tests.cc
Replace scoped_ptr with unique_ptr in webrtc/call/
2016-03-12 14:10:53 +00:00
rampup_tests.cc
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
rampup_tests.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
ringbuffer_unittest.cc
Refactored CL for moving the output to a separate thread.
2016-04-22 19:40:46 +00:00
ringbuffer.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
rtc_event_log2rtp_dump.cc
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
rtc_event_log_helper_thread.cc
Log current time if we stop logging before stop_time_.
2016-07-29 19:38:45 +00:00
rtc_event_log_helper_thread.h
Hibernate the thread if there are no events in the queue. Wake it up when an event is added to the queue.
2016-06-08 14:20:36 +00:00
rtc_event_log_parser.cc
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
rtc_event_log_parser.h
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
rtc_event_log_unittest_helper.cc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
rtc_event_log_unittest_helper.h
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
rtc_event_log_unittest.cc
Generate random rtp packets with RtpPacketToSend instead of RtpSender
2016-08-10 11:23:38 +00:00
rtc_event_log.cc
Reland: Add BWE plot to event log analyzer.
2016-07-29 12:49:04 +00:00
rtc_event_log.h
Reland: Add BWE plot to event log analyzer.
2016-07-29 12:49:04 +00:00
rtc_event_log.proto
Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
2015-11-06 17:00:21 +00:00
transport_adapter.cc
Add a PacketOptions struct to webrtc::Transport.
2015-10-02 10:39:40 +00:00
transport_adapter.h
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
webrtc_call.gypi
Move congestion controller to a separate module.
2016-02-23 12:30:51 +00:00