10 Commits

Author SHA1 Message Date
terelius
84e78f9102 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)

Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.

Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.

BUG=webrtc:5177

Review URL: https://codereview.webrtc.org/1457023002

Cr-Commit-Position: refs/heads/master@{#10965}
2015-12-10 09:51:02 +00:00
Fredrik Solenberg
ea07373a2e Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
BUG=webrtc:5268,webrtc:5273
TESTED=Fixed issues reported by:
find webrtc/audio -type f -name *.cc -o -name *.h | xargs cpplint.py
find webrtc/call -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1483323002 .

Cr-Commit-Position: refs/heads/master@{#10853}
2015-12-01 10:26:46 +00:00
terelius
d66daa2d2f Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
BUG=

Review URL: https://codereview.webrtc.org/1419523004

Cr-Commit-Position: refs/heads/master@{#10542}
2015-11-06 17:00:21 +00:00
terelius
56b1128c8f Change to use local Random object instead of global rand() in the RtcEventLog unit test.
Removed Rand(int low, int high) since that function outputs results that are non-random and/or outside the interval if low is negative.

Added new Uniform(uint32_t, uint32_t) function to replace Rand(int low, int high).

Changed various unit tests to use the new functions.
BUG=

Review URL: https://codereview.webrtc.org/1413053002

Cr-Commit-Position: refs/heads/master@{#10541}
2015-11-06 13:14:01 +00:00
terelius
006d93d3c6 Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
BUG=

Review URL: https://codereview.webrtc.org/1411673003

Cr-Commit-Position: refs/heads/master@{#10531}
2015-11-05 20:02:19 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
terelius
1adce14c87 Old config events are no longer removed from RtcEventLog.
Time to keep old events in buffer is now changeable at runtime.
Added unit test for removing old events from buffer.

Review URL: https://codereview.webrtc.org/1303713002

Cr-Commit-Position: refs/heads/master@{#10302}
2015-10-16 15:51:15 +00:00
Ivo Creusen
301aaed813 Update to the RtcEventLog protobuf to remove the DebugEvent message.
This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.

This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1348113003 .

Cr-Commit-Position: refs/heads/master@{#10221}
2015-10-08 16:07:53 +00:00
pbos
da903eaabb Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1373903003

Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 09:37:18 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00