77e11bbe83
Wire up preferred/nominal_bitrate to stats.
...
Also adds a test that shows that actual_enc_bitrate was not summed
correctly plus fixing it.
Additionally reducing locking when grabbing stats.
BUG=1778
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34319004
Cr-Commit-Position: refs/heads/master@{#8464}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8464 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 16:39:58 +00:00
1ed6224eaf
Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
...
This breaks compilation outside this codebase that needs to have it
removed before.
BUG=4322
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42009004
Cr-Commit-Position: refs/heads/master@{#8432}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8432 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 13:57:43 +00:00
8ad05b7628
Remove dead stats from Video{Sender,Receiver}Info.
...
These stats are neither filled nor plumbed further and might as well be
removed (as proven by how easy they were to remove).
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39219004
Cr-Commit-Position: refs/heads/master@{#8430}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8430 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 13:00:46 +00:00
1d0fa5d352
Add RtcpPacketTypeCounter stats to new API.
...
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
b4987bfc24
Send black frame with previous size when muting.
...
Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.
R=mflodman@webrtc.org , pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/39179004
Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-18 10:13:45 +00:00
e9facf8bb3
Add range checks in a variety of places where the values will subsequently be
...
expected to be 0-127.
BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika
Review URL: https://webrtc-codereview.appspot.com/37759004
Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 20:37:35 +00:00
86196c4f48
Setup encoders inexpensively before first frame.
...
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 21:02:20 +00:00
ff689be3c0
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 11:55:32 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
2b19f06312
Wire up RTT statistics to webrtc::Call.
...
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/32249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:26:09 +00:00
a85307737c
(Auto)update libjingle 81702493-> 81755413
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 09:01:18 +00:00
599e299b9d
cricket::VideoFrame int64 to int64_t.
...
Needed for successful compile of ios arm64.
BUG=3898
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30359004
Patch from Zeke Chin <tkchin@webrtc.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 09:42:57 +00:00
0fb6ad2004
Check if cpu_monitor_ exists before Stop().
...
R=asapersson@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 13:44:29 +00:00
273a414b0e
Report encoded frame size in VideoSendStream.
...
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +00:00
008731868a
Implement settable min/start/max bitrates in Call.
...
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/26199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 14:03:34 +00:00
bdcf38c894
cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class
...
There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc
R=fbarchard@google.com , pbos@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 10:53:00 +00:00
d819803d45
Wire up DSCP support in WebRtcVideoEngine2.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/24249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 14:41:43 +00:00
957e802fe0
Refactor SetDefaultEncoderConfig to work on existing codecs.
...
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/32009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 12:36:11 +00:00
188d3b2245
Enable VP9 video codec support on webrtcvideoengine behind a field trial.
...
BUG=chromium:431285
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 13:21:04 +00:00
a2ef4fe9c3
Prevent a lot of VideoSendStream reconfigures.
...
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 10:54:43 +00:00
82775b1396
Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime.
...
This will allow to plugin VP9 based on a field trial.
R=pbos@webrtc.org , pbos, pthatcher
Review URL: https://webrtc-codereview.appspot.com/27949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 09:37:54 +00:00
19b4741004
Removing unused method GetDefaultVideoEncoderConfig.
...
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 11:16:32 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
88ef632286
Falling back on single-stream on multiple SSRC.
...
Instead of failing, use one stream. Also clamp video min bitrate.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:29:29 +00:00
96a93259b3
Implement external decoder support in WebRtcVideoEngine2.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 14:46:44 +00:00
b7ed7799e7
Implement conference-mode temporal-layer screencast.
...
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 13:08:10 +00:00
3bf3d238c8
Configure A/V sync in WebRtcVideoEngine2.
...
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/23249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 12:59:34 +00:00
776e6f289c
Use external VideoDecoders in VideoReceiveStream.
...
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org , stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
ae694effd8
(Auto)update libjingle 78642371-> 78680406
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 17:37:17 +00:00
efc82c2c73
Implement screencast settings for WebRtcVideoEngine2.
...
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 13:58:00 +00:00
fa553ef605
Set up start bitrate in WebRtcVideoEngine2.
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R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/27789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 11:07:07 +00:00
1ecbe45c7e
(Auto)update libjingle 77689511-> 77696841
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 20:29:28 +00:00
7fe1e03dd6
Wire up external encoders.
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R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 04:25:33 +00:00
3c16d8bd1c
(Auto)update libjingle 77414393-> 77554188
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 06:35:10 +00:00
97abeee282
(Auto)update libjingle 77263371-> 77296420
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 22:24:30 +00:00
