* Remove "dead" and "alive" variables.
* Remove critical section
* Skip synchronizing with the worker thread to verify startup (no need).
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
Also added some TODOs for myself for the ThreadWrapper interface.
I'm removing the HasNoMonitorThread test since it is no longer relevant and ends up checking the wrong thing (ProcessThread - a generic thread type) in the wrong way (parsing a debug log) :) I think it served a purpose some years ago, but things have changed since.
BUG=2902
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37069004
Cr-Commit-Position: refs/heads/master@{#8220}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8220 4adac7df-926f-26a2-2b94-8c16560cd09d
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.
BUG=2822
R=hta@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
> Update StatsCollector's interface in preparation of more changes.
>
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
>
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
>
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
>
> BUG=2822
> R=perkj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36829004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.
Note that both Chrome and the Android implementation use an external signaling thread.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
