Sergey Ulanov ec4f068bcd Style cleanups in RtpSender. ...

- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h

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

Review URL: https://codereview.webrtc.org/2067673004 .

Cr-Commit-Position: refs/heads/master@{#13565}

2016-07-28 22:19:18 +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

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

Style cleanups in RtpSender.

2016-07-28 22:19:18 +00:00

rtc_event_log.cc

Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ )

2016-07-28 16:21:32 +00:00

rtc_event_log.h

Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ )

2016-07-28 16:21:32 +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