22 Commits

Author SHA1 Message Date
Stefan Holmer
13181035bc Reland: Add BWE plot to event log analyzer.
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.

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

Review URL: https://codereview.webrtc.org/2193763002 .

Cr-Commit-Position: refs/heads/master@{#13574}
2016-07-29 12:49:04 +00:00
terelius
59c1ad58e6 Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ )
Reason for revert:
Breaks downstream targets.

Original issue's description:
> Add BWE plot to event log analyzer.
>
> The plot is constructed by actually running the congestion controller with
> the logged rtp headers and rtcp feedback messages to reproduce the same behavior
> as in the real call.
>
> R=phoglund@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/2beea2a8c920000ef19eea20cce397507fc3d5e7
> Cr-Commit-Position: refs/heads/master@{#13558}

TBR=phoglund@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2190013002
Cr-Commit-Position: refs/heads/master@{#13559}
2016-07-28 16:21:32 +00:00
Stefan Holmer
2beea2a8c9 Add BWE plot to event log analyzer.
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.

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

Review URL: https://codereview.webrtc.org/2188033004 .

Cr-Commit-Position: refs/heads/master@{#13558}
2016-07-28 15:42:31 +00:00
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
ivoc
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00
Ivo Creusen
1895526c61 Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.

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

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
tommi
a6219cc3ef FileWrapper[Impl] modifications and actually remove the "Impl" class.
This is a somewhat involved refactoring of this class. Here's an overview of the changes:

* FileWrapper can now be used as a regular class and instances allocated on the stack.
* The type now has support for move semantics and copy isn't allowed.
* New public ctor with FILE* that can be used instead of OpenFromFileHandle.
* New static Open() method.  The intent of this is to allow opening a file and getting back a FileWrapper instance.  Using this method instead of Create(), will allow us in the future to make the FILE* member pointer, to be const and simplify threading (get rid of the lock).
* Rename the Open() method to is_open() and make it inline.
* The FileWrapper interface is no longer a pure virtual interface.  There's only one implementation so there's no need to go through a vtable for everything.
* Functionality offered by the class, is now reduced.  No support for looping (not clear if that was actually useful to users of that flag), no need to implement the 'read_only_' functionality in the class, since file APIs implement that already, no support for *not* managing the file handle (this wasn't used).  OpenFromFileHandle always "manages" the file.
* Delete the unused WriteText() method and don't support opening files in text mode.  Text mode is only different on Windows and on Windows it translates \n to \r\n, which means that files such as log files, could have a slightly different format on Windows than other platforms.  Besides, tools on Windows can handle UNIX line endings.
* Remove FileName(), change Trace code to manage its own path.
* Rename id_ member variable to file_.
* Removed the open_ member variable since the same functionality can be gotten from just checking the file pointer.
* Don't call CloseFile inside of Write.  Write shouldn't be changing the state of the class beyond just attempting to write.
* Remove concept of looping from FileWrapper and never close inside of Read()
* Changed stream base classes to inherit from a common base class instead of both defining the Rewind method. Ultimately, Id' like to remove these interfaces and just have FileWrapper.
* Remove read_only param from OpenFromFileHandle
* Renamed size_in_bytes_ to position_, since it gets set to 0 when Rewind() is called (and the size actually does not change).
* Switch out rw lock for CriticalSection. The r/w lock was only used for reading when checking the open_ flag.

BUG=

Review-Url: https://codereview.webrtc.org/2054373002
Cr-Commit-Position: refs/heads/master@{#13155}
2016-06-15 17:30:18 +00:00
terelius
bea8959687 Hibernate the thread if there are no events in the queue. Wake it up when an event is added to the queue.
BUG=614192

Review-Url: https://codereview.webrtc.org/2035483003
Cr-Commit-Position: refs/heads/master@{#13070}
2016-06-08 14:20:36 +00:00
terelius
43587e32e9 Take ownership and close the platform file even if we fail to start logging.
Review-Url: https://codereview.webrtc.org/2015543002
Cr-Commit-Position: refs/heads/master@{#12944}
2016-05-27 09:22:54 +00:00
isheriff
6f8d686d35 Remove use of RtpHeaderExtension and clean up
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension

The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.

Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.

BUG= webrtc:5895

Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
2016-05-26 18:25:04 +00:00
ivoc
c1513ee1a3 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API.
The caller can set a negative or zero file size to avoid using a limit.
BUG=

Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
2016-05-13 15:30:44 +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
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
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
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
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
ivoc
112a3d81db Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1374253002

Cr-Commit-Position: refs/heads/master@{#10297}
2015-10-16 09:22:23 +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