18 Commits

Author SHA1 Message Date
nisse
037ee92455 Fix test.gyp dependency.
BUG=

Review URL: https://codereview.webrtc.org/1907303002

Cr-Commit-Position: refs/heads/master@{#12487}
2016-04-25 08:39:23 +00:00
terelius
4311ba59d8 Refactored CL for moving the output to a separate thread.
The logging thread is always active. The main thread uses SwapQueues to pass events to the logging thread. The logging thread moves the events to either a RingBuffer history in memory, or to a string which is written to disc.

RtcEventLogImpl constructor takes a clock for easier testing.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1687703002

Cr-Commit-Position: refs/heads/master@{#12476}
2016-04-22 19:40:46 +00:00
solenberg
6021fe2b1e Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
kwiberg
b25345ee3f Replace scoped_ptr with unique_ptr in webrtc/call/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1789903003

Cr-Commit-Position: refs/heads/master@{#11970}
2016-03-12 14:10:53 +00:00
danilchap
69e59e619a [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
rtcp::RawPacket is rtc::Buffer, it had no extra functionality.
rtc::Buffer is a movable class - no point to wrap it into rtc::scoped_ptr
change is large, but straightforward:
  rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
  ->Buffer() replaced with .data()
  ->Length() replaced with .size()

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1696203002

Cr-Commit-Position: refs/heads/master@{#11649}
2016-02-17 11:11:50 +00:00
terelius
429c345b02 Fixes a bug which incorrectly logs incoming RTCP as outgoing.
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1571283002

Cr-Commit-Position: refs/heads/master@{#11336}
2016-01-21 13:42:10 +00:00
danilchap
34ed2b95a5 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1544983002

Cr-Commit-Position: refs/heads/master@{#11288}
2016-01-18 10:43:38 +00:00
Peter Boström
e2976c87f7 Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.

This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.

The change also removes gtest_disable.h as an unused include from many
other files.

BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1547343002 .

Cr-Commit-Position: refs/heads/master@{#11150}
2016-01-04 21:44:16 +00:00
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