cfc319be1d
Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
...
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..
(This time from my @webrtc account.)
Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org ,stefan@webrtc.org ,philipel@google.com ,asapersson@webrtc.org ,mflodman@webrtc.org ,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}
TBR=sprang@webrtc.org ,stefan@webrtc.org ,philipel@google.com ,asapersson@webrtc.org ,mflodman@webrtc.org ,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1431283002
Cr-Commit-Position: refs/heads/master@{#10581}
2015-11-10 15:17:26 +00:00
0be8f1d347
Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
...
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}
TBR=sprang@webrtc.org ,stefan@webrtc.org ,philipel@google.com ,asapersson@webrtc.org ,mflodman@webrtc.org ,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1438543002
Cr-Commit-Position: refs/heads/master@{#10578}
2015-11-10 13:31:22 +00:00
77ccfb4d16
Work on flexible mode and screen sharing.
...
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.
Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1328113004
Cr-Commit-Position: refs/heads/master@{#10572}
2015-11-10 10:19:20 +00:00
49e196af40
Remove VideoFrameType aliases for FrameType.
...
No longer used in Chromium, so these can now be removed.
BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1415693002 .
Cr-Commit-Position: refs/heads/master@{#10390}
2015-10-23 13:58:27 +00:00
22993e1a0c
Unify FrameType and VideoFrameType.
...
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.
BUG=webrtc:5042
R=magjed@webrtc.org , mflodman@webrtc.org , henrik.lundin@webrtc.org , solenberg@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1371043003
Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
5d0379da2c
Remove kSkipFrame usage.
...
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
2015-10-06 12:05:03 +00:00
a9455ab235
Integration of VP9 packetization.
...
Supports running 1 spatial and 1-3 temporal layers in non-flexible mode.
BUG=webrtc:4148, webrtc:4168, chromium:500602
TBR=mflodman
Review URL: https://codereview.webrtc.org/1211353002
Cr-Commit-Position: refs/heads/master@{#9665}
2015-07-31 13:10:16 +00:00
ab00404571
VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument
...
BUG=484432
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53379004
Cr-Commit-Position: refs/heads/master@{#9135}
2015-05-05 09:37:17 +00:00
54d072ea20
Add CVO support to video_coding layer.
...
CVO is, instead of rotating frame on the capture side, to have renderer rotate the frame based on a new rtp header extension.
The change includes
1. encoder side needs to pass this from raw frame to the encoded frame.
2. decoder needs to copy it from rtp packet (only the last packet of a frame has this info) to decoded frame.
R=mflodman@webrtc.org
TBR=stefan@webrtc.org
BUG=4145
Review URL: https://webrtc-codereview.appspot.com/46429006
Cr-Commit-Position: refs/heads/master@{#8767}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8767 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 21:55:37 +00:00
2ec560606b
Add H.264 packetization.
...
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
34c5da6b5e
Cleaned up logging in video_coding.
...
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 14:08:35 +00:00
71f055fb41
Add send frame rate statistics callback
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 15:09:27 +00:00
f1e807c0e5
Removing FrameForStorage
...
R=pwestin@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2142004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 22:34:41 +00:00
822fbd8b68
Update talk to 50918584.
...
Together with Stefan's http://review.webrtc.org/1960004/ .
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
aa4d96a134
Revert r4301
...
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
a4407329d4
Include files from webrtc/.. paths in video_coding/.
...
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
...
rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +00:00
418443c531
Remove operator overloading from RTPFragmentationHeader.
...
Instead supply a CopyFrom() method.
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/972004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3158 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 19:17:23 +00:00
14b43beb7c
Move src/ -> webrtc/
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
