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1090 Commits

Author SHA1 Message Date
glaznev@webrtc.org
bc35703694 Add a method to remove an existing renderer from the internal list of Android renderers. 
BUG=4290
R=jiayl@webrtc.org, mquiros@google.com

Review URL: https://webrtc-codereview.appspot.com/36089004

Cr-Commit-Position: refs/heads/master@{#8320}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 23:23:47 +00:00
glaznev@webrtc.org
bc40324d9c Merge fixes and changed for Android AppRTCDemo from internal repo. 
- Rename AppRTCDemoActivity to CallActivity and move UI controls
to a fragment.
- Add option to enable/disable statistics.
- Move peer connection and video constraints from URL to peer
connection client.
- Variable renaming.

R=jiayl@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33299004

Cr-Commit-Position: refs/heads/master@{#8319}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 23:05:04 +00:00
pbos@webrtc.org
f4c10d24dc Always use DeliverI420Frame in WebRtcVideoEngine. 
Moves native_handle() path to DeliverI420Frame and CHECKs that
DeliverFrame is not being used anymore.

R=magjed@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/38019004

Cr-Commit-Position: refs/heads/master@{#8312}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 10:20:38 +00:00
glaznev@webrtc.org
44ae4c8b07 Support using VP9 video codec in AppRTCDemo. 
- Add peer connection Java API to initialize field trial string.
- Add setting option to select VP8 or Vp9 as default video codec.
- Minor code clean up and allowing 720p portrait encoding.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39899004

Cr-Commit-Position: refs/heads/master@{#8303}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 23:26:41 +00:00
pbos@webrtc.org
0d852d5c27 Use VideoReceiveStream as an ExternalRenderer. 
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.

Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.

R=stefan@webrtc.org
TBR=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/34169004

Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 15:15:24 +00:00
andresp@webrtc.org
53d9012faf Clean kForever from basictypes and move it to the interfaces that actually have it. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33269004

Cr-Commit-Position: refs/heads/master@{#8296}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 14:19:39 +00:00
pbos@webrtc.org
8cf9bdb3fa Remove USE_WEBRTC_DEV_BRANCH. 
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.

R=bjornv@webrtc.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39849004

Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 10:17:12 +00:00
guoweis@webrtc.org
6c930c7183 Cleanup: unify rotation to be enum based instead of int for degree. 
Split from https://webrtc-codereview.appspot.com/37029004/

BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8257

Committed: https://code.google.com/p/webrtc/source/detail?r=8276

Committed: https://code.google.com/p/webrtc/source/detail?r=8277

Review URL: https://webrtc-codereview.appspot.com/37129004

Cr-Commit-Position: refs/heads/master@{#8288}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 01:29:45 +00:00
guoweis@webrtc.org
0c7ec770ff Cleanup: unify rotation to be enum based instead of int for degree. 
Split from https://webrtc-codereview.appspot.com/37029004/

BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8257

Committed: https://code.google.com/p/webrtc/source/detail?r=8276

Review URL: https://webrtc-codereview.appspot.com/37129004

Cr-Commit-Position: refs/heads/master@{#8277}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 21:01:47 +00:00
guoweis@webrtc.org
110443aaac Cleanup: unify rotation to be enum based instead of int for degree. 
Split from https://webrtc-codereview.appspot.com/37029004/

BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8257

Review URL: https://webrtc-codereview.appspot.com/37129004

Cr-Commit-Position: refs/heads/master@{#8276}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 20:00:46 +00:00
perkj@webrtc.org
9baa9ca399 Add libjingle_peerconnection_so.so to Java test dependencies. 
This fix a problem where the Java test is not dependent on the so file.

BUG=4275
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33239004

Cr-Commit-Position: refs/heads/master@{#8270}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 16:09:20 +00:00
magjed@webrtc.org
4b320cf214 Revert "Cleanup: unify rotation to be enum based instead of int for degree." 
Reason for revert:
Compile error on bots - A subclass of cricket::VideoFrame still uses old GetRotation return type.

BUG=4145
TBR=guoweis,stefan,pthatcher

This reverts commit 3e733a43f5.

Review URL: https://webrtc-codereview.appspot.com/34159004

Cr-Commit-Position: refs/heads/master@{#8265}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 12:58:46 +00:00
guoweis@webrtc.org
57ac2c84dd Default destination used by c line should be IPv4 only to avoid parsing error in legacy client. 
Make sure the IP family overwrites the preference of candidates. Also,
make sure only UDP is used as default destination.

BUG=4269
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36009004

Cr-Commit-Position: refs/heads/master@{#8258}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-06 00:45:43 +00:00
guoweis@webrtc.org
3e733a43f5 Cleanup: unify rotation to be enum based instead of int for degree. 
Split from https://webrtc-codereview.appspot.com/37029004/

BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37129004

Cr-Commit-Position: refs/heads/master@{#8257}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-05 23:40:43 +00:00
glaznev@webrtc.org
f6932297e7 Fix Android video renderer to support video frames 
with stride > width.

Recent libvpx update generates output video frames with stride
value greater than width, which was not supported by Android OpenGL
video renderer (Android GLES2 doesn't have GL_UNPACK_ROW_LENGTH
to provide stride information for buffer in glTexImage2D call).

Fix it by implementing native frame copying for Java
VideoRenderer.I420Frame implementation.

BUG=4248
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40639004

Cr-Commit-Position: refs/heads/master@{#8252}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-05 17:30:17 +00:00
bjornv@webrtc.org
cc64a9cc4f voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric 
As of r8230 (https://webrtc-codereview.appspot.com/39739004/) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.

This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine

BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41749004

Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-05 12:53:24 +00:00
pthatcher@webrtc.org
877ac765ad Cleanup and prepare for bundling. 
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38909004

Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 22:03:41 +00:00
bjornv@webrtc.org
520a69e8ea Revert 8238 "Add RefCounting for TransportProxies" 
Failing on Mac64_Debug

> Add RefCounting for TransportProxies
> 
> BUG=1574
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/37869004

TBR=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37159004

Cr-Commit-Position: refs/heads/master@{#8243}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 12:46:13 +00:00
bjornv@webrtc.org
c5f697135e Revert 8237 "Cleanup and prepare for bundling." 
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.

> Cleanup and prepare for bundling.
> 
> - Add a GetOptions function. Needed for eventual bundle testing to
>   confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
> 
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/39699004

TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34959004

Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 10:22:43 +00:00
decurtis@webrtc.org
e2506670a4 Add RefCounting for TransportProxies 
BUG=1574
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37869004

Cr-Commit-Position: refs/heads/master@{#8238}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-03 23:19:23 +00:00
pthatcher@webrtc.org
af01d93aa2 Cleanup and prepare for bundling. 
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39699004

Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-03 23:14:18 +00:00
decurtis@webrtc.org
322a564f49 Fix datachannel stats id and timestamp. 
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel.

Adds a timestamp to the data channel reports.

Implements unit tests to verify that the timestamp is set correctly.

BUG=1805
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33119004

Cr-Commit-Position: refs/heads/master@{#8236}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-03 22:10:13 +00:00
pkasting@chromium.org
0e81fdf5d2 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. 
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40569004

Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 23:54:40 +00:00
pkasting@chromium.org
19f3f71c98 Fix apparent typo: int -> char. 
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.

BUG=none
TEST=none
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40529004

Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 19:44:42 +00:00
pkasting@chromium.org
026b892e72 Using << on an int8_t or uint8_t will output a character rather than a number. 
Places that do this need to cast to int to get the desired behavior.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40579004

Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-30 19:54:19 +00:00
pkasting@chromium.org
005b6fffe6 Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails. 
BUG=none
TEST=none
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39649004

Cr-Commit-Position: refs/heads/master@{#8222}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-30 19:42:17 +00:00
pbos@webrtc.org
5e161616b1 Remove CPU monitor from WebRtcVideoEngine2. 
CPU adaptation is based on timings done inside webrtc, not actual CPU
values anymore. This code has never been wired up and is causing flakes
on at least valgrind, but possibly also on actual platforms.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34089004

Cr-Commit-Position: refs/heads/master@{#8221}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-30 15:31:26 +00:00
tommi@webrtc.org
aef0779dab Rewrite ThreadWindows. 
* 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
 2015-01-30 15:06:44 +00:00
braveyao@webrtc.org
8820ac7cc4 peerconnectin_server: missing comma in sprintfn() in r8128 
BUG=4244
TEST=Manual Test
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37079004

Cr-Commit-Position: refs/heads/master@{#8213}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-30 09:58:45 +00:00
pbos@webrtc.org
50fe359eb6 Add tracing for slow paths in new video API. 
Allows tracking what actually takes time in SetRemoteDescription and
SetLocalDescription.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38809004

Cr-Commit-Position: refs/heads/master@{#8202}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 12:33:42 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID. 
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
 2015-01-29 12:14:13 +00:00
magjed@webrtc.org
a26f511dd2 Remove frame copy in ViEExternalRendererImpl::RenderFrame 
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.

BUG=1128,4227
R=mflodman@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8136

Review URL: https://webrtc-codereview.appspot.com/36489004

Cr-Commit-Position: refs/heads/master@{#8199}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 11:45:43 +00:00
braveyao@webrtc.org
a742cb1f37 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off. 
BUG=3872
TEST=Manual Test
R=jiayl@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36989004

Cr-Commit-Position: refs/heads/master@{#8193}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 04:23:39 +00:00
pkasting@chromium.org
e7a4a12f83 Add arraysize() macro from Chromium, and make use of it in a few places. 
This not only shortens some test code, it makes it more robust against changing
the lengths of the arrays later and forgetting to update the length constants
(which bit me).

BUG=none
TEST=none
R=hta@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34829004

Cr-Commit-Position: refs/heads/master@{#8191}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 21:37:13 +00:00
honghaiz@google.com
a67ca1a3bb Only report the first rtp packet because it indicates the media has started flowing. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37829004

Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:48:40 +00:00
tkchin@webrtc.org
36401aba62 Update GAE API paths for join/leave. 
BUG=4221
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33069004

Cr-Commit-Position: refs/heads/master@{#8174}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 21:35:16 +00:00
magjed@webrtc.org
fc5ad95fec Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139 
Link to original CL: https://review.webrtc.org/36909004/

R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227

Review URL: https://webrtc-codereview.appspot.com/39669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 09:57:01 +00:00
glaznev@webrtc.org
8501ee632b Support VP8 HW decoding on devices with Exynos codec. 
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 23:07:19 +00:00
glaznev@webrtc.org
82415e395f Update AppRTCDemo to use renamed GAE messages. 
BUG=4221
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 22:22:50 +00:00
tkchin@webrtc.org
7519de519e Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..." 
> Remove frame copy in ViEExternalRendererImpl::RenderFrame
> 
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
> 
> BUG=1128
> R=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36489004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 21:20:41 +00:00
tkchin@webrtc.org
0f98844749 Revert 8139 "Implement elapsed time and capture start NTP time e..." 
> Implement elapsed time and capture start NTP time estimation.
> 
> These two elements are required for end-to-end delay estimation.
> 
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36909004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 21:17:38 +00:00
jiayl@webrtc.org
dacdd9403d Reland r7980: 
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
 2015-01-23 17:33:34 +00:00
pbos@webrtc.org
ad3ee2c46b Implement elapsed time and capture start NTP time estimation. 
These two elements are required for end-to-end delay estimation.

BUG=1788
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 14:55:00 +00:00
kjellander@webrtc.org
a02d76845f Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness 
Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.

Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.

BUG=4219
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 14:34:52 +00:00
magjed@webrtc.org
182ea46fac Remove frame copy in ViEExternalRendererImpl::RenderFrame 
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.

BUG=1128
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 11:50:13 +00:00
tommi@webrtc.org
586f2eda0d Change GetStreamBySsrc to not copy StreamParams. 
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
 2015-01-22 23:00:41 +00:00
jlmiller@webrtc.org
b40c7bb53c Change sprintf use in talk samples to snprintf 
BUG=2301
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 18:49:06 +00:00
asapersson@webrtc.org
cfd82dfc11 Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 
Prepares for adding FEC bytes to the StreamDataCounter.

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

Review URL: https://webrtc-codereview.appspot.com/37579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 09:39:59 +00:00
jiayl@webrtc.org
cceb166a3f Fix a use-after-free when sending queued messages is aborted for blocked channel. 
BUG=4187
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 00:55:10 +00:00
tommi@webrtc.org
4fb7e25843 Update StatsReport and by extension StatsCollector to reduce data copying. 
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
 2015-01-21 11:36:18 +00:00