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158 Commits

Author SHA1 Message Date
pbos
d6fc47ea95 Remove base channel for video receivers. 
BUG=webrtc:1695
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1251163002

Cr-Commit-Position: refs/heads/master@{#9624}
 2015-07-23 13:58:36 +00:00
asapersson
d89920b74a Add resolution and fps stats to histograms: 
- "WebRTC.Video.InputWidthInPixels"
- "WebRTC.Video.InputHeightInPixels"
- "WebRTC.Video.SentWidthInPixels"
- "WebRTC.Video.SentHeightInPixels"
- "WebRTC.Video.ReceivedWidthInPixels"
- "WebRTC.Video.ReceivedHeightInPixels"
- "WebRTC.Video.RenderFramesPerSecond"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1228393008

Cr-Commit-Position: refs/heads/master@{#9611}
 2015-07-22 13:52:03 +00:00
Jelena Marusic
cd6702282a Define Stream base classes 
BUG=webrtc:4690

Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.

R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1226123005 .

Cr-Commit-Position: refs/heads/master@{#9591}
 2015-07-16 07:30:20 +00:00
pbos
8fc7fa798f Base A/V synchronization on sync_labels. 
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.

BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1181653002

Cr-Commit-Position: refs/heads/master@{#9586}
 2015-07-15 15:03:04 +00:00
pbos
ba8c15b857 Merge methods for configuring NACK/FEC/hybrid. 
BUG=webrtc:1695
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1226143013

Cr-Commit-Position: refs/heads/master@{#9580}
 2015-07-14 16:36:37 +00:00
Åsa Persson
1aff095b6c Moved check for native frame to VideoReceiveStream::FrameCallback. 
Stats for decoded framerate will now also be updated if the frame is backed by a texture.

BUG=webrtc:4722
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53569004

Cr-Commit-Position: refs/heads/master@{#9389}
 2015-06-08 07:26:40 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame. 
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
 2015-05-30 00:21:56 +00:00
Peter Boström
2251d6e174 Remove ViESender. 
Registers transport on construction removing the need for ViESender as a
hop and removing a potential deadlock by removing RegisterSendTransport.

BUG=1695, 2999
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57449004

Cr-Commit-Position: refs/heads/master@{#9309}
 2015-05-28 12:10:48 +00:00
Peter Boström
def39883f0 Configure default render delay as 10 ms. 
BUG=chromium:488395
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56419005

Cr-Commit-Position: refs/heads/master@{#9296}
 2015-05-27 15:59:21 +00:00
Peter Boström
5a3ebd761c Revert "Remove default encoder/decoders." 
This reverts commit 78ae00eea29850bd3bd608287d8b057c88e91b42 due to perf
regressions. Reverting during investigation to figure out root causes.

BUG=chromium:491112
R=henrik.lundin@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56449004

Cr-Commit-Position: refs/heads/master@{#9283}
 2015-05-26 09:44:14 +00:00
Peter Boström
78ae00eea2 Remove default encoder/decoders. 
This path is not used, senders/receivers already disable default coders.

BUG=1695
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54449004

Cr-Commit-Position: refs/heads/master@{#9245}
 2015-05-21 07:56:17 +00:00
Peter Boström
45553aefac Remove VideoEngine interface usage from new API. 
Instantiates ProcessThread/ChannelGroup inside Call instead of using
VideoEngine or ViEBase. This removes the need for ViEChannelManager,
ViEInputManager and other ViESharedData completely.

Some interface headers are still referenced due to external interfaces
being defined there. Upon interface removal these will be moved to
implementation headers.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50849005

Cr-Commit-Position: refs/heads/master@{#9160}
 2015-05-08 11:54:39 +00:00
Peter Boström
5cb9ce4c74 Remove ViECodec usage in VideoSendStream. 
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also inlining the necessary parts of SetSendCodec
done previously in ViECodecImpl.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46129004

Cr-Commit-Position: refs/heads/master@{#9136}
 2015-05-05 13:16:40 +00:00
Peter Boström
52ef9d7738 Stop IncomingVideoStream on delete. 
Fixes race between VideoReceiveStream destruction and pending
IncomingVideoStream frames.

BUG=
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45259004

Cr-Commit-Position: refs/heads/master@{#9084}
 2015-04-24 16:06:33 +00:00
Peter Boström
c4188fd3c7 Use IncomingVideoStream in VideoReceiveStream. 
Decouples VideoReceiveStream further from webrtc/video_engine/ as well
as most of webrtc/modules/video_render/ resulting in a simpler setup.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50749004

Cr-Commit-Position: refs/heads/master@{#9080}
 2015-04-24 13:15:40 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8. 
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
 2015-04-21 12:25:42 +00:00
Peter Boström
76c53d36bc Remove ViE interface usage from VideoReceiveStream. 
References channels and underlying objects directly instead of using
interfaces referenced with channel id. Channel creation is still done as
before for now.

BUG=1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46849004

Cr-Commit-Position: refs/heads/master@{#8958}
 2015-04-09 12:35:46 +00:00
Peter Boström
23914fe756 Reject RTP one-byte extension ID 0. 
Only accept local identifiers in the range 1-14 inclusive.

BUG=1788, chromium:471328
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50549004

Cr-Commit-Position: refs/heads/master@{#8900}
 2015-03-31 13:08:13 +00:00
pbos@webrtc.org
2b4ce3a501 Convert webrtc/video/ abort/assert to CHECK/DCHECK. 
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.

BUG=1756
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42879004

Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 13:13:15 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer. 
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:51:50 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
9e4e524f38 Use an external-only VideoRenderModule in Call. 
The default render module instantiated from inside VideoEngine if none
exists instantiates platform-specific code. Call only uses external
rendering, so this is an unneccessary overhead.

BUG=1667
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39069004

Cr-Commit-Position: refs/heads/master@{#8346}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 10:48:55 +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
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
pbos@webrtc.org
32e8528581 Log configs when creating video streams in Call. 
Adds VideoReceiveStream::Config::ToString and logs configs in both
Call::CreateVideoSendStream and Call::CreateVideoReceiverStream.

R=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/41519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 10:09:39 +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
pbos@webrtc.org
5570769210 Remove the last getters from VideoReceiveStream stats. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/32899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:45:03 +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
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
pbos@webrtc.org
09cc686c8b Delete VideoReceiveStream channels in destructor. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/31909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:48:15 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
26c0c41a06 Network up/down signaling in Call. 
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:17:12 +00:00
pbos@webrtc.org
6ae48c6609 Make VideoSendStream/VideoReceiveStream configs const. 
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.

CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.

This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.

R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260

Review URL: https://webrtc-codereview.appspot.com/20409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 10:49:19 +00:00
pbos@webrtc.org
a5c8d2c9b3 Rename Start/Stop in Video{Send,Receive}Streams. 
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.

R=mflodman@webrtc.org
BUG=3227

Review URL: https://webrtc-codereview.appspot.com/12329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 11:13:21 +00:00
pbos@webrtc.org
2a03498825 Implement FEC support in VideoReceiveStream. 
Added an FEC end-to-end test. NACK+FEC is probably working but not yet tested
as the test for it must introduce packet delays as the underlying API prefers
NACK over FEC if RTT is low.

BUG=3174
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:21:45 +00:00
pbos@webrtc.org
f577ae9eac Remove internal codecs from VideoSendStream. 
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.

For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.

Removes Call::GetVideoCodecs().

Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.

BUG=2854,2992
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-19 08:43:57 +00:00
sprang@webrtc.org
9510e53cc0 Make VideoReceiveStream::GetStats() const. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 15:32:45 +00:00
sprang@webrtc.org
09315705b9 Wire up statistics in video receive stream of new API 
This CL includes Call tests that test both send and receive sides.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 12:06:29 +00:00
sprang@webrtc.org
d9b9560ee5 Drop early packets when not sending in TransportAdapter. 
Particularly, suppress periodic RTCP packets before
VideoSendStream.StartSending() or VideoReceiveStream.StartReceiving() have been called, respectively.

RTCP packets are sent periodically, by the Process thread, for every ViE channel even those not sending.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 13:03:02 +00:00
pbos@webrtc.org
c279a5d72c Wire up RTX in VideoReceiveStream. 
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.

BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 09:30:53 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00
mflodman@webrtc.org
b429e516a9 cpplint cleaning new API and its implementation files. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 09:46:22 +00:00
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293. 
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 12:24:44 +00:00
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." 
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 23:57:54 +00:00
mflodman@webrtc.org
92c2793154 Adding REMB to receive stream configuration, the send side will always 
react to incoming REMB for now.

Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.

TEST=See above.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 16:36:28 +00:00
mflodman@webrtc.org
f3973e81d5 Make sure channels in the same call are in the same channel group. 
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 09:40:45 +00:00
wu@webrtc.org
a9890800e0 Update talk to 58127566 together with 
https://webrtc-codereview.appspot.com/5309005/.

R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 00:21:03 +00:00