| 822bdf978435b8eba9343ea96e9a9bc54b9c7df0 |
|
11-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Remove cricket::VideoEncoderConfig.
BUG=webrtc:5332
R=noahric@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1512853007 .
Cr-Commit-Position: refs/heads/master@{#10991}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1387149ad1669365ac05278bf779a407bec08a4e |
|
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 9d69c3f4d99240c27d997c37950b561605d403bd |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Return a copy of the supported RTP header extensions instead of a reference.
This also renames the method to better reflect what it does.
BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1486123002 .
Cr-Commit-Position: refs/heads/master@{#10910}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| bd13838ccc87f94d1e951bcf780979622b020359 |
|
21-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1457653003
Cr-Commit-Position: refs/heads/master@{#10734}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| be57983f4bd875c39a229bab5112b32dad004057 |
|
10-Nov-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Rename Maybe to Optional
And add examples of good and bad usage to the documentation.
R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1432553007 .
Cr-Commit-Position: refs/heads/master@{#10588}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 102c6a61bc0b42dc0956d013530fc0213b7e881b |
|
30-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Replace rtc::cricket::Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1d8a506405734d0cef9653704b036ca4f1388960 |
|
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
|
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 34fbfff068bf46d27812fb8fd531aea889a5feaf |
|
24-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoMediaChannel::SetRender().
Was a no-op in current implementation.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1334793003 .
Cr-Commit-Position: refs/heads/master@{#10059}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 61e933eac7673feb2f8663c3e71e503b714b350f |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364083002
Cr-Commit-Position: refs/heads/master@{#10045}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| b071a19019a0a2173cc139c960d6ef6946a1c581 |
|
17-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1327933002 .
Cr-Commit-Position: refs/heads/master@{#9973}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 709ed67c38d0a942f3bf3e68e337cc27a27bc353 |
|
15-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1269863005 .
Cr-Commit-Position: refs/heads/master@{#9939}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1dd98f321920c1442dd5b3f791ea0fca133c2756 |
|
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 66f43392a31ac566565e910246ef496fcbbafb04 |
|
09-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1324853003
Cr-Commit-Position: refs/heads/master@{#9904}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 874ca3af5b163e1b3fd8802171e44ee252557842 |
|
21-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Don't do reconfiguration if recv codec order/preference changes
Adding 'ReceiveCodecsHaveChanged' method that will determine if codecs
HAVE changed, irrespective of order and preference.
Review URL: https://codereview.webrtc.org/1291763003
Cr-Commit-Position: refs/heads/master@{#9748}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 048e80cacab128280c37c3e0574873cde8544410 |
|
19-Aug-2015 |
tommi <tommi@webrtc.org> |
Revert of Revert "Remove CpuMonitor and related, unused, code." (patchset #1 id:1 of https://codereview.webrtc.org/1287913004/ )
Reason for revert:
(retrying with my webrtc account...)
The reason for reverting is: Re-landing the change that removes the CpuMonitor class after having fixed the build issue in Chromium..
Original issue's description:
> Revert "Remove CpuMonitor and related, unused, code."
>
> This reverts commit 1a24012680f25440aa1d117373df2af14cdc2fc1.
>
> TBR=tommi@webrtc.org,pthatcher@webrtc.org
> BUG=
>
> This breaks
> http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/20148/steps/compile/logs/stdio
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/a472e968c95fb14e63ec42f453551d0967573ea8
TBR=pthatcher@webrtc.org,guoweis@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1290033005
Cr-Commit-Position: refs/heads/master@{#9733}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| a472e968c95fb14e63ec42f453551d0967573ea8 |
|
19-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Remove CpuMonitor and related, unused, code."
This reverts commit 1a24012680f25440aa1d117373df2af14cdc2fc1.
TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=
This breaks
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/20148/steps/compile/logs/stdio
Review URL: https://codereview.webrtc.org/1287913004 .
Cr-Commit-Position: refs/heads/master@{#9730}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1a24012680f25440aa1d117373df2af14cdc2fc1 |
|
18-Aug-2015 |
Tommi <tommi@webrtc.org> |
Remove CpuMonitor and related, unused, code.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1298953002 .
Cr-Commit-Position: refs/heads/master@{#9727}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| c2ee2c86f905991a8cd05ee1f35bea105b41e4e0 |
|
08-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229283003 .
Cr-Commit-Position: refs/heads/master@{#9690}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| c27d89fdc6b33846ff06e8ca8bd03119d05c6530 |
|
16-Jul-2015 |
qiangchen <qiangchen@chromium.org> |
Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 259bd2034c3d3ee7f2dc4d481e9bf896a3a4d6ef |
|
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Report ssrc_groups in GetStats().
This was already available in the stats struct, just not filled in.
BUG=4720
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47329004
Cr-Commit-Position: refs/heads/master@{#9308}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 3548dd21542c7b3f2c4680c6a6d86b0d719bd008 |
|
22-May-2015 |
Peter Boström <pbos@webrtc.org> |
Set local SSRCs on receivers added before senders.
Addresses bug where a receiver would report SSRC 1 even though the
endpoint has sending streams.
BUG=4678
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51099004
Cr-Commit-Position: refs/heads/master@{#9262}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 9a416bd14ee225d8f1a1ada627a1dd7daf275032 |
|
22-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51879004
Cr-Commit-Position: refs/heads/master@{#9258}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 4d71edef45afa38b3f68b6af0519ac0f21d327df |
|
19-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software encoding.
Permits falling back to software encoding for unsupported resolutions.
BUG=chromium:475116, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46279004
Cr-Commit-Position: refs/heads/master@{#9227}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 7252a2ba8035c4128917a9558a3e34fc9dbe7c44 |
|
18-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software decoding.
Permits falling back to software decoding for unsupported resolutions in
bitstreams.
BUG=4625, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46269004
Cr-Commit-Position: refs/heads/master@{#9209}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 67c9df79828991c5aab96b9253ae4e7ba7ed508e |
|
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base NACK on send codecs.
Addressing discrepancy where NACK used to be set from send codecs in
WebRtcVideoEngine(1), and before this change, from recv codecs in
WebRtcVideoEngine2. This should address that NACK might be sent even if
the remote side does not support it.
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53409004
Cr-Commit-Position: refs/heads/master@{#9171}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 126c03ea02d8a99bfa3d1e6d6fe04516183d31af |
|
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base decision to send REMB on send codecs.
Fixes bug where Chromium would send REMB even though the remote party
doesn't announce support for it (because it was based on local codec
settings instead of remote ones).
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54389004
Cr-Commit-Position: refs/heads/master@{#9170}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 4b60c73e74d62beff484b7f54d8f3267cb66274f |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 81ea54eaac82b36b7208a02fd37a469d7d0bd9d0 |
|
07-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove WebRtcVideoEngine.
Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until
build files in Chromium have been modified.
BUG=1695,4566
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48339004
Cr-Commit-Position: refs/heads/master@{#9148}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 143cec1cc68b9ba44f3ef4467f1422704f2395f0 |
|
28-Apr-2015 |
Erik Språng <sprang@google.com> |
Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.
BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46069004
Cr-Commit-Position: refs/heads/master@{#9093}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| ee0b00e8a9cc2d8f4578912a389dee92ac020ee9 |
|
22-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent recv-stream reconfig on identical codecs.
Receive streams seem to be reconfigured with identical codecs when
another stream is removed. Preventing this reconfiguration makes sure
that existing streams don't report stats during teardown when the stream
is still supposed to be running.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44249004
Cr-Commit-Position: refs/heads/master@{#9059}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 56d50288e0f5df75cddc3798b8e01cdb75f25c92 |
|
14-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove SignalCaptureStateChange from MediaEngine.
It's no longer used by anything.
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/48069004
Cr-Commit-Position: refs/heads/master@{#8994}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 77f0e3f7b6a0664661dc295eb235c543b8091554 |
|
13-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48039004
Cr-Commit-Position: refs/heads/master@{#8990}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| e7b221f4760af10e29cb4c501e758cc3518f628b |
|
13-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove deadlock in WebRtcVideoEngine2.
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.
R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065
Review URL: https://webrtc-codereview.appspot.com/50679004
Cr-Commit-Position: refs/heads/master@{#8985}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 9bfe3daf7349b62647997ced9389baa8ab043afe |
|
10-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Remove i420_video_frame.h header.
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 64c1e8cda5cb4db85c5c296bf2f6a8181af7de9d |
|
02-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8917}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 31331cfd2d3d17958942b67190c8b943c05b084f |
|
01-Apr-2015 |
Minyue <minyue@webrtc.org> |
Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae.
Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.
Review URL: https://webrtc-codereview.appspot.com/47909004
Cr-Commit-Position: refs/heads/master@{#8911}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1b1c15cad16de57053bb6aa8a916079e0534bdae |
|
01-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| dfd53fe26b013d0948024a38eec6fbc31c29a094 |
|
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Raise streams for SetMaxSendBitrates above 2000k.
Fixes b=AS effectively not setting bitrates above 2000k.
BUG=1788,4469
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47839004
Cr-Commit-Position: refs/heads/master@{#8882}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| d6f4c25eedcfd502920f1b2a24744badd9da80be |
|
26-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject streams reusing simulcast or RTX SSRCs.
BUG=1788, chromium:470122, chromium:470856
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42919004
Cr-Commit-Position: refs/heads/master@{#8868}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 8296ec518b2659de922668bfe0db57e71eb17e74 |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44779004
Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| afdd5dd372d69be7244a3d90d70de9d5ecd60eb9 |
|
12-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.
Reason for revert: The thread checker in Chromium that crashed has been fixed now.
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/40319004
Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| b218ff553148b9a26c82e3b3a46d626c4438cedd |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.
Reason for revert: Fails on Chromium FYI content_browsertests
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/47529004
Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 370a72cc3ff928099c6ec6766659ed12155b74df |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy from cricket::VideoFrame to I420VideoFrame
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42249004
Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| a2a6fe66a39797ea61a04d80ce3afc494d850bfc |
|
06-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Reconfigure default streams on AddRecvStream.
Makes sure RTX can be used for streams that have received early media
before being properly configured.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46499004
Cr-Commit-Position: refs/heads/master@{#8634}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 058b1f17ac43b1fe69a8c18aaa7999ba88733dfd |
|
04-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.
BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47379004
Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| f1f0d9a4cd53f4eacbf791cb7317612fa7382a45 |
|
02-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEngine::SetVoiceEngine.
Instead enforcing that a voice engine is set on construction. Apart from
simplifying the class this permits tracing to be set up in the
constructor without worrying about racing sets from SetVoiceEngine
later.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44489004
Cr-Commit-Position: refs/heads/master@{#8555}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8555 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 9a4410e9934578e84cc129b978a29e151d957994 |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement adaptation stats in WebRtcVideoEngine2.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42489004
Cr-Commit-Position: refs/heads/master@{#8510}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8510 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| b4987bfc24e1e755a6c54053d09a58d1e72228bb |
|
18-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 86196c4f481d7f515e54806988f763169e8b9206 |
|
16-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 5e161616b17900c06809e7275afca96363d44ad5 |
|
30-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| fc5ad95fecc5ddc7d98dcfbac1c4e75a7814253f |
|
27-Jan-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 0f988447496e5d656d52bea279c8511d3569cb11 |
|
23-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| ad3ee2c46bf502a18847229d42dd081c9e753c70 |
|
23-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| f1c8b905204bc7a6c74271ead038f5d80d8d3eed |
|
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| f18fba2f7b3d1fad7b7b38a9a5dc281bef06c50e |
|
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 2b19f0631233488e891d9db0d170b637dc8fc464 |
|
11-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| a85307737cc9ea3e79b86daf96d455fca4ad1bb4 |
|
10-Dec-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 81702493-> 81755413
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 008731868a09e2fe01da53733a612dc24761f791 |
|
25-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| a2ef4fe9c331e7668b9e8ff64ce5141a535a5f21 |
|
07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 19b47410044aecac20f3f46a4d207018fc466e2e |
|
06-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
|
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 96a93259b361f4b03080a188d781b0835cf4edaf |
|
03-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
|
| efc82c2c734171faba9e400ff60a114e7af2ebcc |
|
27-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| fa553ef6053b20f3768d5fe4314e8c993648bf0a |
|
20-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set up start bitrate in WebRtcVideoEngine2.
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 1ecbe45c7e4c9142896cb2810d699558518f4f28 |
|
14-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77689511-> 77696841
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 7fe1e03dd6da66401010119734245f114bf06645 |
|
14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up external encoders.
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
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 3c16d8bd1c0a3eea94a6678497eae0cf8e7f0187 |
|
13-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77414393-> 77554188
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 97abeee2825ac93b62397feea74d0ad02d42540d |
|
09-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77263371-> 77296420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 575d126a3d4a4bf6d43ea07189ac201f6bfe0798 |
|
08-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Protect send_/recv_streams_ in WebRtcVideoEngine2.
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/22959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 42684be21b255e2b07eb154e6a2807fa2226167e |
|
03-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up CPU adaptation in WebRtcVideoEngine2.
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.
BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| d60d79a14594cbc8266e4a50391ddbe64ed491f0 |
|
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Thread annotation of rtc::CriticalSection.
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.
This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.
R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 38344ed2806c8fed60d67d280ca44c32e36707c0 |
|
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 0a2087a7110e2455ce68f2c85068df5ae447508f |
|
23-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Skeleton for registering external encoders/decoders.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 83f95ba9a645099df5e19a91030029181d766b40 |
|
22-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove engine-level SetOptions.
Already removed in WebRtcVideoEngine.
R=andresp@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| bbe0a8517d7f9da7aa779bff77cdbb70df358437 |
|
19-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 992febb9978d2ded1a2c3c8a42ea18ee071ca3ae |
|
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74873066-> 74873164
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 818b7b3ac982e9e1f579904c5f160103da046dcf |
|
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74825084-> 74825992
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| c4175b9fdf7d23eb58a044ff39e2b096f9091995 |
|
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set resolution based on incoming VideoFrames.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/17269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 9f341283f64d9b905c3883fd23988eb3d5fdcb8f |
|
02-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEngine::default_codec_format().
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/24399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7029 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| b648b9d85c5d07b0866ef45f5be587f71b0849b4 |
|
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove test constructor in WebRtcVideoEngine2.
Removes the need for ::Construct().
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 3740d741068698baf987b1ced5ea485378e16d04 |
|
23-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73927658-> 73927775
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| ef8bb8d9b0bca0b1fd1ddb0a17df665e9dfaf9ad |
|
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure that muting muted streams succeeds.
We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 2c0fb05f1683b7a721072bdd93501b8afe164b9a |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73221069-> 73222930
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| afb554f404d68e6f3ca5395216f776169370713d |
|
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move default-recv-channels to a separate class.
BUG=1788,3099
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 6f48f1bf68a10669c9bcd81262c1a98ed2a8d462 |
|
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement encoder options in WebRtcVideoEngine2.
Implementing default options to enable denoising by default and wiring
up encoder settings to propagate VP8 settings.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| e6f84ae8a602ce78733d20b280ce32198e7ecef5 |
|
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial WebRtcVideoEngine2::GetStats().
Also forward-declaring and moving WebRtcVideoRenderer out of header.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| d1ea06b3d5adab352741df5092c56b20f3e1a74f |
|
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restart VideoReceiveStreams in WebRtcVideoEngine2.
Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.
Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 5301b0f1fce9478dfa56476e174332a1d67b053a |
|
17-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move additional state into WebRtcVideoSendStream.
Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 587ef60056ff0e301a95a9eb8231fb0cae6b69b1 |
|
16-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement RTP extension support in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| d41eaeb7cded2b2cda83f53aa320cf18e2d07380 |
|
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69005149-> 69049090
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
|
| 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/talk/media/webrtc/webrtcvideoengine2.h
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| 0d523eea831e616c415c61765127ed5eb17e5f11 |
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05-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove static initializer from WebRtcVideoEngine2.
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
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| b5a22b14648c53874b4b76368a1a2271d985e875 |
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13-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.
BUG=
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
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| 17911dca8099707b5c050741a108b95b79a4da66 |
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12-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66798415-> 66813165
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
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| d266a2020f9e86a787eada77d458ee75426d68af |
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12-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial wiring of new webrtc API in libjingle.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.h
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