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484 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
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
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
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency 
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 21:36:13 +00:00
sprang@webrtc.org
ff9462eb54 Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. 
Tests are flaky on tsan, disabling for now.

BUG=4135
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 12:06:35 +00:00
pbos@webrtc.org
f1c8b90520 Remove WebRtcVideoEncoderFactory2. 
This interface is no longer required and just adds complexity.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/33009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 17:29:27 +00:00
pbos@webrtc.org
f18fba2f7b Implement SimulcastEncoderAdapter support. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/37589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:26:23 +00:00
henrik.lundin@webrtc.org
8315d7de85 Remove dual stream functionality in VoiceEngine 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:07:26 +00:00
mflodman@webrtc.org
b4e5d1b34e Remove RTX SSRC when deleting the default receive stream. 
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 15:07:07 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
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
 2015-01-09 20:22:46 +00:00
pbos@webrtc.org
2a169640a3 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>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
decurtis@webrtc.org
2ead571fb6 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. 
BUG=3996
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 19:18:01 +00:00
pbos@webrtc.org
59062d5aef Rename SendAndReceiveH264SvcQqvga to VP8 instead. 
This looks like it's been incorrect for a while, this test configures
VP8 in QQVGA.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 19:21:18 +00:00
pbos@webrtc.org
bab79951ca Convert FileMediaEngineTest to use more expects. 
Allows pinpointing more precisely where a failure occurs.

BUG=4144
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 18:01:29 +00:00
kjellander@webrtc.org
07c83a1385 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) 
In https://webrtc-codereview.appspot.com/35669004/ the wrong
define was used (OS_WIN only exists in Chromium code).

BUG=4135
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 10:36:53 +00:00
kjellander@webrtc.org
d95435c17a Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win 
These tests have turned out to be flaky on Windows.

BUG=4135
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-06 11:01:35 +00:00
pbos@webrtc.org
c37e72e890 Make setting identical RTP extensions a no-op. 
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.

BUG=1788,4077
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 18:51:13 +00:00
pbos@webrtc.org
896888b7e4 Remove min bitrate from simulcast streams. 
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/38529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 15:40:56 +00:00
sprang@webrtc.org
46d4d29a75 Add field trial for screenshare bitrates when using temporal layers. 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 15:19:35 +00:00
braveyao@webrtc.org
086c8d5a02 Use a temporary buffer to scale a screencast in OnFrameCaptured 
BUG=3903
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/23909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-22 05:46:42 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API. 
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:33:17 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

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

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
pthatcher@webrtc.org
e2b7585bc2 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:09:08 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
magjed@webrtc.org
e575e9c40f Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h 
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 02:44:30 +00:00