| 796cfaf7f76aa740cc7f4bb2c94f88637e475324 |
|
10-Dec-2015 |
perkj <perkj@webrtc.org> |
Add VideoCodec::PreferDecodeLate
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.
Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.
Review URL: https://codereview.webrtc.org/1428293003
Cr-Commit-Position: refs/heads/master@{#10974}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| d1590b2571c4cb33416e14c92e4f2dfed42ec3d4 |
|
09-Dec-2015 |
mflodman <mflodman@webrtc.org> |
Lint clean video/ and add lint presubmit check.
BUG=webrtc:5316
Review URL: https://codereview.webrtc.org/1507643004
Cr-Commit-Position: refs/heads/master@{#10953}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 7623ce4aeb9130c937ba5836490cbb3a35679e79 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.
Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}
BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1510183002 .
Cr-Commit-Position: refs/heads/master@{#10948}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 8237abf563bf4782ee104408b53cc8e55ce44518 |
|
08-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}
TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695
Review URL: https://codereview.webrtc.org/1507903005
Cr-Commit-Position: refs/heads/master@{#10937}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 03ef053202bc5d5ab43460eebf5403232f157646 |
|
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1506773002 .
Cr-Commit-Position: refs/heads/master@{#10926}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 187db63fdfbfb462a722b685e4a1faf40632d257 |
|
01-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoReceiveStream deregister of decoders.
Also doing some simplifications inside video_coding. No CHECKs added,
since they appear to have introduced breakages in downstream tests.
Overall reducing the number of potential ways a decoder could possibly
be set null. Removing deregistration of external decoders should also
give a quicker shutdown time since that may attempt to register
internal decoders.
BUG=chromium:563299
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1483423002 .
Cr-Commit-Position: refs/heads/master@{#10858}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 521af4e344678ce9dcf996341af6ba8056e1e147 |
|
27-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove duplicate decoders in BitrateEstimatorTest.
Multiple decoders were used for the same payload type in this test case,
causing CHECK failures when configuring.
BUG=webrtc:5249
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1484443003 .
Cr-Commit-Position: refs/heads/master@{#10825}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 795dbe4e0fa78333d52fe39edb08a0b13a281c34 |
|
27-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove RegisterExternal{De,En}coder error codes.
Also adds a RTC_CHECK in VideoReceiveStream that verifies that decoders
aren't null, since this will attempt to deregister a codec which would
previously fail with an obscure stack trace not indicating what actually
was wrong.
BUG=webrtc:5249
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1479793002 .
Cr-Commit-Position: refs/heads/master@{#10821}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 444682acf9804c5fcbddaded9e900ba3cc6921fc |
|
25-Nov-2015 |
qiangchen <qiangchen@chromium.org> |
Remove frame time scheduing in IncomingVideoStream
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 43edf0ffb91a50e2efa01c7befe4d188a7e30ea2 |
|
21-Nov-2015 |
stefan <stefan@webrtc.org> |
Require negotiation to send transport cc feedback over RTCP.
BUG=4312
Review URL: https://codereview.webrtc.org/1452883002
Cr-Commit-Position: refs/heads/master@{#10735}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 0a41893e36148642d8f37fc91738e42eca47fef8 |
|
13-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Remove BitrateController dependency fromVideoReceiveStream.
I have another CL moving REMB from CongestonController to Call, then
I'll remove CongestoinController from this class too.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1442003002 .
Cr-Commit-Position: refs/heads/master@{#10632}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 0e7e259ebd69993bb5670a991f43aa1b06c9bf9e |
|
13-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 415d2cd7454d93b3727fce9147090a24e4c3ccba |
|
26-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for video.
BUG=webrtc:5118
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1415413004 .
Cr-Commit-Position: refs/heads/master@{#10403}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 0c478b3d75be3c026e68f03a11cb558c3655c926 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Rename ChannelGroup to CongestionController and move to webrtc/call/.
BUG=webrtc:5079
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1419803002 .
Cr-Commit-Position: refs/heads/master@{#10358}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| e37870297fc45f1253dff4b1c59c85a50d2a8a97 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
ChannelGroup cleanup.
Move CallStats to Call, EncoderStateFeedback to VideoSendStream and
remove last ViEChannel dependency from ChannelGroup.
BUG=webrtc:5079
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1418613002 .
Cr-Commit-Position: refs/heads/master@{#10355}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 86b016027d2d27c62fedd108354a2b1274118ae3 |
|
21-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Add stats for average QP per frame for VP8 (for received video streams):
"WebRTC.Video.Decoded.VP8.Qp"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1340623002
Cr-Commit-Position: refs/heads/master@{#10349}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| eff0fc677589d356c458d324af4a81781054e119 |
|
20-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Adding missing stats class registration, lost in #10298.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1408233004 .
Cr-Commit-Position: refs/heads/master@{#10334}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 0dbf0090a961c5e5fb7362937108337564b4a91f |
|
19-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Remove the video channel id completely.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1412143002
Cr-Commit-Position: refs/heads/master@{#10324}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| a20de2030f7f3a3c5e252ccc76a467109f5a93dc |
|
19-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Move ownership of receive ViEChannel to VideoReceiveStream.
This CL changes as little as possible and I'll follow up later with
ownership of the other members in ChannelGroup.
The next step is to remove the id used for channels.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1411723002
Cr-Commit-Position: refs/heads/master@{#10318}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| a2f30deea342896ee40cc4d90567f091efbe0fc9 |
|
15-Oct-2015 |
pbos <pbos@webrtc.org> |
Log Call {audio, video} stream deletions.
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1400333002
Cr-Commit-Position: refs/heads/master@{#10286}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 65220a70a38ffe252b587775c5e9104606ab7c2c |
|
14-Oct-2015 |
noahric <noahric@chromium.org> |
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.
Review URL: https://codereview.webrtc.org/1394573004
Cr-Commit-Position: refs/heads/master@{#10276}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| e23e737177cf5d131a6d4a4d229aa513c5270a59 |
|
08-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Disable pacer disabling.
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| da903eaabbb6c6830efcafc3c2ade1d36f511e43 |
|
02-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1373903003
Cr-Commit-Position: refs/heads/master@{#10143}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 6b8d3551681f40b880507cecc88f478a12383cc7 |
|
24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc
BUG=webrtc:4173
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1362303002 .
Cr-Commit-Position: refs/heads/master@{#10052}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
|
23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29
Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}
TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1362923002 .
Cr-Commit-Position: refs/heads/master@{#10029}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| ef165eefc79cf28bb67779afe303cc2365885547 |
|
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1338203003
Cr-Commit-Position: refs/heads/master@{#10012}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
|
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 05cfcd34693d86de7a1a481f071eae561361588a |
|
07-Sep-2015 |
ivica <ivica@webrtc.org> |
Full stack graphs
Updating full stack test to optionally save metadata for each frame and save it
to a file with given filename (controlled from the new full_stack_samples
executable).
Adding a Python script that reads the output generated by full stack test
and plots the graph(s).
Review URL: https://codereview.webrtc.org/1289933003
Cr-Commit-Position: refs/heads/master@{#9874}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| f42376c60111edba6f29060bf3dd79e75d8dbb97 |
|
28-Aug-2015 |
pbos <pbos@webrtc.org> |
Wire up currently-received video codec to stats.
BUG=webrtc:1844, webrtc:4808
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1315413002
Cr-Commit-Position: refs/heads/master@{#9810}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d |
|
28-Aug-2015 |
solenberg <solenberg@webrtc.org> |
Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1273363005
Cr-Commit-Position: refs/heads/master@{#9807}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
|
03-Aug-2015 |
sprang <sprang@webrtc.org> |
Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| d6fc47ea9585fa405d8617ccc0b7704d2c9e8a0a |
|
23-Jul-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| d89920b74a173b7bf80c6760908a382c095a66cc |
|
22-Jul-2015 |
asapersson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| cd6702282a49448adda470934f4bd9e6181cab22 |
|
16-Jul-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 8fc7fa798f7a36955f1b933980401afad2aff592 |
|
15-Jul-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| ba8c15b857c0f341d9c1e02d41b6ccd56f9f1030 |
|
14-Jul-2015 |
pbos <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 1aff095b6cc39d843a3d85d09de599cfcff0e35e |
|
08-Jun-2015 |
Åsa Persson <asapersson@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 4765070b8d6f024509c717c04d9b708750666927 |
|
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 2251d6e17438e1a085ff4f88ad19de513214bec1 |
|
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| def39883f00c25525dfd34c3cee92b428e54b9e5 |
|
27-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 5a3ebd761cb2789aa63e4ae7c0035234a3e3b5f3 |
|
26-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 78ae00eea29850bd3bd608287d8b057c88e91b42 |
|
21-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 45553aefacb797818da83ccef1c3679a8aa0fc7f |
|
08-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 5cb9ce4c746867a02e7d37358f63e1a7c11ef262 |
|
05-May-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 52ef9d77386e8d3e40c8d3d07b93f06950aede72 |
|
24-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| c4188fd3c74688264621393fc622cb81c042c1ac |
|
24-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| e62202fedf57b74cc263246c0586ee353978caf8 |
|
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 76c53d36bc455fe89ca1f860d5171633198fe907 |
|
09-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 23914fe756903353eae13fffc868d2c84f51f06f |
|
31-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 2b4ce3a501b8d679f84c1ad10317dea5c78fa595 |
|
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| fdd10579496123c9a7fdc0bf185e2a26a12ed340 |
|
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 9e4e524f3894c7eeee8a53bd3cbf21d27b5efc8c |
|
12-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| f4c10d24dc8f1c3ce6859644077d7df6fb678dcd |
|
10-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 0d852d5c27a759fe7aadc500bd7b3cadfae3deb8 |
|
09-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 32e852858101c3565cfc79cdda9310a3336d95a0 |
|
15-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 8f27fcce79584378da97f0d84574564799e138d6 |
|
09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 2a169640a3225a559f926fe74f1fe1af239e191f |
|
09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 55707692105a4765f8f321ec7c30a1034d03d23a |
|
19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| ce4e9a356200170abcdd44ff2af95f87a6781b8e |
|
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 09cc686c8bc4b5e766d38e6860ac52aa886e2436 |
|
04-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 3bf3d238c8c4578e444e5a601684db68c79a29ca |
|
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 776e6f289c7396a1143b8b36b03f88b08ac8cba3 |
|
29-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 26c0c41a06d77af54df547169d952a21319dea8c |
|
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 6ae48c660934784b4df56ab1ac99402ce3745e9f |
|
06-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| a5c8d2c9b39a2d20fead2147e60ed0cd6d62019c |
|
24-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 2a034988257074183615615f0cc7f59f3411a21c |
|
08-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| f577ae9eac9822380ea6f0fb953cf383d0ec5374 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 9510e53cc06b0aa5be2be78fbab375216067eea2 |
|
07-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 09315705b9caf3bff455e3515b9bd99492a7c3e3 |
|
07-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| d9b9560ee50c236efcb690ee479021b415f7dfd4 |
|
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| c279a5d72c885b1a1737018ee26dc7c0475a38bf |
|
24-Jan-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| efaeda0c76fbf9a58c44931d525348ab59dd52b0 |
|
20-Jan-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| b429e516a98d2dee0c57d3263f6d21633939b564 |
|
18-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 5ab756703ea32f2c2ff9878d6eae628c7380bc14 |
|
16-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 41e2615e020311172b937f527c13d9e090437eca |
|
15-Dec-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 341e91441aaa9c2c5a638082c3ee4530aa21612c |
|
14-Dec-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 92c2793154334e6931a86a68d1196b870c406d54 |
|
13-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| f3973e81d5aa7e4f1d6b5abdfe3a6dc53a32840c |
|
13-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| a9890800e078105f21f0a21358ee59a0b3736af6 |
|
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 2018269dc3a1c1bb01c946583ca0750ae0db68e3 |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5274 "Update talk to 58113193 together with https://webrt..."
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
>
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| a129b6cd132788a931b47da3370ae473673f320d |
|
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58113193 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@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| b613b5ab2b03041942f04fd892e2ad5a4f9de027 |
|
03-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set local SSRC for VideoReceiveStream.
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.
BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 4070935f4fb5b9fb2df246d7073fe0ba7e350791 |
|
26-Nov-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement and test EncodedImageCallback in new ViE API.
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| d29d4e9c08f50e470eaa239957c5859996856d0c |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Deliver I420VideoFrames from VideoRender module.
Performance issue and simplicity, this implementation skips conversion
to VideoEngine's frame format and then back again to I420VideoFrame.
BUG=2526
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 53c85735256dc7d540deb0a5e2bbb2f2821c4bd4 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename video streams' start/stop methods.
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| b082ade3db118113bba284d0f8fd32901371a2a0 |
|
18-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Hook up audio/video sync to Call.
Adds an end-to-end audio/video sync test.
BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|
| 16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
|
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_receive_stream.cc
|