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14d5dbe5b35c560e978dab0c02c48cb16d28273d
platform-external-webrtc/webrtc/call
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ivoc 14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" 
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +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
Auto pause video streams based on encoder target bitrate.
2016-07-01 11:04:10 +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
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +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
Implementing auto pausing of video streams.
2016-06-09 15:21:30 +00:00
rampup_tests.h
Replace scoped_ptr with unique_ptr in webrtc/call/
2016-03-12 14:10:53 +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
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +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
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
rtc_event_log.cc
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
rtc_event_log.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +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