| ee37de3c13c5fe4b397d909918e9f980dc8184c5 |
|
23-Nov-2015 |
sprang <sprang@webrtc.org> |
Add screenshare perf tests with lossy links
This is a re-land of https://codereview.webrtc.org/1409513005/
Fingers crossed, the problems previously seen have been resolved by
https://codereview.webrtc.org/1412233003/
BUG=
Review URL: https://codereview.webrtc.org/1409993011
Cr-Commit-Position: refs/heads/master@{#10751}
/external/webrtc/webrtc/video/full_stack.cc
|
| 464c0878d915aed3ea9abc799ad08bf53791a0fd |
|
13-Nov-2015 |
philipel <philipel@webrtc.org> |
Rename screenshare test.
Renamed the test to reflect what is actually tested. What the old test
did I don't know since there has never been possible to use screenshare
with two temporal layers in VP9.
BUG=chromium:554515
Review URL: https://codereview.webrtc.org/1441693002
Cr-Commit-Position: refs/heads/master@{#10631}
/external/webrtc/webrtc/video/full_stack.cc
|
| cfc319be1d6afec77bd41eeb70d3e7886dd524db |
|
10-Nov-2015 |
philipel <philipel@webrtc.org> |
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}
/external/webrtc/webrtc/video/full_stack.cc
|
| 0be8f1d347bdb171462df89c2a4c69b3f3eb7519 |
|
10-Nov-2015 |
terelius <terelius@webrtc.org> |
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}
/external/webrtc/webrtc/video/full_stack.cc
|
| 77ccfb4d16c148e61a316746bb5d9705e8b39f4a |
|
10-Nov-2015 |
philipel <philipel@webrtc.org> |
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}
/external/webrtc/webrtc/video/full_stack.cc
|
| ce4aef16eec96862199e89b6d3ffe059558ac2c0 |
|
02-Nov-2015 |
sprang <sprang@webrtc.org> |
Adding support for simulcast and spatial layers into VideoQualityTest
This is a re-land of https://codereview.webrtc.org/1353263005/
which was reverted because of perf-regressions. Changes since that CL:
* Change LayerFilteringTransport to send a padding packet instead of
dropping it for data that should be filtered out. This prevents
confusion due to changed sequence numbers.
* Changed timing of stats poller thread in VideoAnalyzer. Startup was
racy wrt initializion of send_stream_.
* Minor formatting issues.
PERF NOTE: This change will affect some performance numbers slightly.
In particular, {encode_frame_rate, encode_time_ms,
encode_usage_percent, media_bitrate_bps} will change due to timing
of the measurements.
BUG=
R=pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1412233003
Cr-Commit-Position: refs/heads/master@{#10483}
/external/webrtc/webrtc/video/full_stack.cc
|
| 65a036780bc6ed19e0f52b6f16a56b8b4eb07086 |
|
19-Oct-2015 |
sprang <sprang@webrtc.org> |
Revert of Add screenshare perf tests with lossy links (patchset #1 id:1 of https://codereview.webrtc.org/1409513005/ )
Reason for revert:
Reverting to see of this resolves build bot failures (Nexus 7.2, especially debug) which now seems to sometimes fail to start tests altogether.
Original issue's description:
> Add screenshare perf tests with lossy links
>
> BUG=
>
> Committed: https://crrev.com/4af0f1a098bc908cfe825981b825687673d5134a
> Cr-Commit-Position: refs/heads/master@{#10290}
TBR=pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1415603002
Cr-Commit-Position: refs/heads/master@{#10322}
/external/webrtc/webrtc/video/full_stack.cc
|
| 4af0f1a098bc908cfe825981b825687673d5134a |
|
15-Oct-2015 |
sprang <sprang@webrtc.org> |
Add screenshare perf tests with lossy links
BUG=
Review URL: https://codereview.webrtc.org/1409513005
Cr-Commit-Position: refs/heads/master@{#10290}
/external/webrtc/webrtc/video/full_stack.cc
|
| 7a975f75e7fa7a9335411ef22b6687f78f7b297f |
|
12-Oct-2015 |
sprang <sprang@webrtc.org> |
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
Reason for revert:
Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works.
Original issue's description:
> Adding support for simulcast and spatial layers into VideoQualityTest
>
> The CL includes several changes:
> - Adding flags describing the streams and spatial layers.
> - Reorganizing the order of the flags, to make them easier to maintain.
> - Adding a member .params_ to VideoQualityAnalyzer.
> (instead of passing it to every member function manually)
> - Updating VideoAnalyzer to support simulcast.
> (select appropriate ssrc and fix timestamps which are sometimes increased by 1)
> - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
> Changing to first read bitrates and resolution ratios from the flags, if specified.
> If not specified, reverting to the old code are setting the values automatically.
> - Changing the parameters in LayerFilteringTransport, replacing
> xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
>
> Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e
> Cr-Commit-Position: refs/heads/master@{#10215}
TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1397363002
Cr-Commit-Position: refs/heads/master@{#10252}
/external/webrtc/webrtc/video/full_stack.cc
|
| 87f83a9a27d657731ccb54025bc04ccad0da136e |
|
08-Oct-2015 |
ivica <ivica@webrtc.org> |
Adding support for simulcast and spatial layers into VideoQualityTest
The CL includes several changes:
- Adding flags describing the streams and spatial layers.
- Reorganizing the order of the flags, to make them easier to maintain.
- Adding a member .params_ to VideoQualityAnalyzer.
(instead of passing it to every member function manually)
- Updating VideoAnalyzer to support simulcast.
(select appropriate ssrc and fix timestamps which are sometimes increased by 1)
- VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
Changing to first read bitrates and resolution ratios from the flags, if specified.
If not specified, reverting to the old code are setting the values automatically.
- Changing the parameters in LayerFilteringTransport, replacing
xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
Review URL: https://codereview.webrtc.org/1353263005
Cr-Commit-Position: refs/heads/master@{#10215}
/external/webrtc/webrtc/video/full_stack.cc
|
| 7bd242e53dacc22eaddbe27a74bdd30d672e57db |
|
06-Oct-2015 |
ivica <ivica@webrtc.org> |
Enabling screensharing tests for Android
This CL runs the screensharing tests 5 times, and none of the Android trybots failed:
https://codereview.webrtc.org/1377663003/
Therefore, we can now probably enable the tests.
BUG=chromium:513170
Review URL: https://codereview.webrtc.org/1389573004
Cr-Commit-Position: refs/heads/master@{#10179}
/external/webrtc/webrtc/video/full_stack.cc
|
| 1356ba5e6ceb7513137b4052f0c2f9edddb11d5e |
|
22-Sep-2015 |
ivica <ivica@webrtc.org> |
Fixing target_bitrate_bps for a FullStackTest
While refactoring, I incorrectly set the target bitrate for one of the tests to
500k instead of 2000k.
This does not fix all the perf regressions.
BUG=534220
Review URL: https://codereview.webrtc.org/1356123002
Cr-Commit-Position: refs/heads/master@{#10008}
/external/webrtc/webrtc/video/full_stack.cc
|
| 5d6a06c1d29a2061bcf4b321ffceab477a404d51 |
|
17-Sep-2015 |
ivica <ivica@webrtc.org> |
Refactoring full stack and loopback tests
Refactoring full stack, video and screenshare tests to use the same code basis
for parametrization and initialization. This patch is done on top of recently
commited full stack graphs CL https://codereview.webrtc.org/1289933003/, but
virtually no changes have been made to full_stack_plot.py nor to the VideoAnalyzer
in full stack, except moving it to video_quality_test.cc.
Also, full_stack_samples.cc (build target) was removed and replaced with
-output_filename and -duration cmdline arguments in video_loopback and
screenshare_loopback.
The important things to review:
- video_quality_test.h
Is the structure of Params good? (examples of usage can be found in
full_stack.cc, video_loopback.cc and screenshare_loopback.cc)
- video_quality_test.cc
Is the initialization correct? The case for using Analyzer and using local
renderer are different, can they be further merged?
- webrtc_tests.gypi
Reproducing the different bitrate settings the full stack and loopback tests had
was a little bit tricky. To support both simultaneously, I added BitrateConfig
to the Params struct, as well as separate start_bitrate and target_bitrate flags
for loopback tests.
Note: Side-by-side diff for video_quality_test.cc compares that file directly
with the old full_stack.cc, so changes to VideoAnalyzer are clearly visible.
Note: Recent CL I've committed added -num_temporal_layers and -sl_discard_threshold
args to loopback tests. This was removed here. Support for streams and SVC
will be added in a CL following this one.
Review URL: https://codereview.webrtc.org/1308403003
Cr-Commit-Position: refs/heads/master@{#9969}
/external/webrtc/webrtc/video/full_stack.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/full_stack.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/full_stack.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/full_stack.cc
|
| 6ee69aa94c8d3d84d6b30462e38004a465af0a7e |
|
03-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Add scrolling screenshare test to full_stack perf tests.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1298613004 .
Cr-Commit-Position: refs/heads/master@{#9850}
/external/webrtc/webrtc/video/full_stack.cc
|
| 2c27430545098f17bc716335a8677a9d21cecac0 |
|
31-Aug-2015 |
Erik Språng <sprang@webrtc.org> |
Print some output in long perf tests, to keep them alive
At least Android try bots seem to have a timeout that will forcibly shut
down the executable if no output has been observed for 60s. Since full
stack test typically run for 60s we need to output give some to avoid
racy shutdown.
BUG=chromium:513170
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1288453003 .
Cr-Commit-Position: refs/heads/master@{#9822}
/external/webrtc/webrtc/video/full_stack.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/full_stack.cc
|
| d6b243f5f636a5685a8258bbb4b536a172752ee9 |
|
11-Aug-2015 |
ivica <ivica@webrtc.org> |
Enabling screensharing perf test.
It should work now as the packet limit in the jitter buffer has been increased.
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1272153002
Cr-Commit-Position: refs/heads/master@{#9700}
/external/webrtc/webrtc/video/full_stack.cc
|
| 62cde2c38cca543cc86e6c6d4d0e9539fa33b9ba |
|
31-Jul-2015 |
ivica <ivica@webrtc.org> |
Disabling VP9 perf test
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1258853004
Cr-Commit-Position: refs/heads/master@{#9668}
/external/webrtc/webrtc/video/full_stack.cc
|
| 028cf488284735f949ff1edf88a5b9bf731eb184 |
|
30-Jul-2015 |
ivica <ivica@webrtc.org> |
Added FullStack performance test for screensharing with VP9
Review URL: https://codereview.webrtc.org/1215113003
Cr-Commit-Position: refs/heads/master@{#9657}
/external/webrtc/webrtc/video/full_stack.cc
|
| b21fd94ece9236b651fcd0f04d1fefd43fbe0dee |
|
23-Jul-2015 |
Peter Boström <pbos@webrtc.org> |
Temporarily disable ScreenshareSlides on Android.
BUG=chromium:513170
TBR=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1246123004 .
Cr-Commit-Position: refs/heads/master@{#9620}
/external/webrtc/webrtc/video/full_stack.cc
|
| 085856cd35c3ec7d5b73d6788465868bbd99ae26 |
|
22-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Extend full stack tests with more stats
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1216613002 .
Cr-Commit-Position: refs/heads/master@{#9612}
/external/webrtc/webrtc/video/full_stack.cc
|
| 4b91bd08979fcfb191cdae27ad24936beefce735 |
|
26-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Move frame input (ViECapturer) to webrtc/video/.
Renames ViECapturer to VideoCaptureInput and initializes several
parameters on construction instead of setters.
Also removes an old deadlock suppression.
BUG=1695, 2999
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53559004.
Cr-Commit-Position: refs/heads/master@{#9508}
/external/webrtc/webrtc/video/full_stack.cc
|
| 2c4c9148191a10c0e82c9a209d454c6b1ebbaf20 |
|
24-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
In screenshare mode, suppress VP8 bitrate overshoot and increase quality
This change includes several improvements:
* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers
A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.
BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1193513006.
Cr-Commit-Position: refs/heads/master@{#9495}
/external/webrtc/webrtc/video/full_stack.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/full_stack.cc
|
| f2f828374c3ee1e1834c72bb27eaae88ef67bb40 |
|
01-May-2015 |
Peter Boström <pbos@webrtc.org> |
Use rtc::CriticalSection in webrtc/video/.
Removes heap allocation from CriticalSection creation.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50839004
Cr-Commit-Position: refs/heads/master@{#9126}
/external/webrtc/webrtc/video/full_stack.cc
|
| 23fba1ffa0079f70744a83bcf4e85501dc226013 |
|
29-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51749004
Cr-Commit-Position: refs/heads/master@{#9114}
/external/webrtc/webrtc/video/full_stack.cc
|
| 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/webrtc/video/full_stack.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/full_stack.cc
|
| 38492c5b6fbb615159fa32b9cc24cd887295573b |
|
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Re-land 8810 "- Add a SetPriority method to ThreadWr..."
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
>
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> >
> > BUG=
> > R=magjed@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/44729004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/48609004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50459005
Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 90a1cb463092c5189b1a69837731a3395d79f61c |
|
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8810 "- Add a SetPriority method to ThreadWrapper"
Seeing if this is causing roll issues.
> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
>
> BUG=
> R=magjed@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44729004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48609004
Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| b6817d793fa647ec77aaaaf74df82a94e46632bb |
|
20-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
- Add a SetPriority method to ThreadWrapper
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44729004
Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 443ad403f5f3550c54669c01d4b584f5fc8e4512 |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove FullStackTest frame pointer handles.
Simplifies code, speculative fix for a DCHECK crash in ForemanCifPlr5.
BUG=4451
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45809005
Cr-Commit-Position: refs/heads/master@{#8803}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8803 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 361981faa86668cd9b20a2837d0b166fc024cd9b |
|
19-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use scoped_ptr for ThreadWrapper::CreateThread.
BUG=
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45799004
Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| af612d5e0769571544952cbe55e675748afa9bdd |
|
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 2056ee3e3c7683ae4b2c4b12da99c3105c4f46a9 |
|
16-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
This reverts commit r8731.
Reason for revert: Breakes Chromium FYI bots.
TBR=hbos, tommi
Review URL: https://webrtc-codereview.appspot.com/40359004
Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 2dc5fa69b2baef2ece158c9e1285516087faaa53 |
|
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40299004
Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 86639737b83d8877abc4810100e30a8af863189d |
|
13-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove thread id from ThreadWrapper::Start().
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| d7452a016812ab1de69c3d7a53caca5b06c64990 |
|
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| bcead305a2f27c30c72c6a3824fdf12f4b83c2eb |
|
06-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 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/webrtc/video/full_stack.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 343096ac03bfb0e1cadbcead29ce925522c12d5b |
|
23-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Fix incorrect rtx config in full_stack tests.
BUG=4326
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40819006
Cr-Commit-Position: refs/heads/master@{#8455}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8455 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| dc77d7447e06a6448634c720970feea732443e69 |
|
20-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness.
BUG=4326
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37359004
Cr-Commit-Position: refs/heads/master@{#8442}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8442 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 131bea89d6f3742e649be84c91f8fd6c43b62d28 |
|
18-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Offline screenshare quality test, plus loopback.
BUG=4171
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34109004
Cr-Commit-Position: refs/heads/master@{#8408}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.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/full_stack.cc
|
| a5d29fcd5978e5e49f7a568f21a2862c384fd95b |
|
10-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Add unit to dropped frames.
Missing unit causes less dropped frames to be reported as a regression
and not an improvement.
R=stefan@webrtc.org
BUG=chromium:429206
Review URL: https://webrtc-codereview.appspot.com/25139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7666 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| b35b136480a0577befb1facc9edb630d62725777 |
|
20-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make avg_{psnr,ssim}_threshold_ const.
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| c216b9aeaf33347d068ac5d8bb2e57d62753e1f0 |
|
14-Oct-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Add a packet loss full stack test to the new API.
Remove all full stack tests for the old API.
BUG=3750
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 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/webrtc/video/full_stack.cc
|
| 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/webrtc/video/full_stack.cc
|
| ab071daab89462db77158e637ba059dba8c9ece7 |
|
18-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common
Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests
GN changes:
- Not many since there is almost no test definitions.
Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.
Re-enable android tests by reverting 7026 (some tests left disabled).
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| ab990ae43a2b84b103cb3c50bc38502375c13e68 |
|
17-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.
BUG=3070
TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 307d3dbdeed71d42edf38d3828081b11a5a416fb |
|
11-Sep-2014 |
henrikg@webrtc.org <henrikg@webrtc.org> |
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| b420191743fc135222c862deeaa4cf9dec249fe3 |
|
09-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Expose VideoEncoders with webrtc/video_encoder.h.
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.
BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 6f729e8a74a4990ca2560607cbc9907cdfaf0401 |
|
02-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable video_engine_tests and webrtc_perf_tests on Android.
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| b8e9e44eac26ab50c07161c55643e1e442927709 |
|
09-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add full stack test cases with a fake network pipe.
R=pbos@webrtc.org
BUG=1872
Review URL: https://webrtc-codereview.appspot.com/20889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 62bafae6618fe3aefbd18657062abc98a40c3375 |
|
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| be9d2a45499d87f3b04e644fc173b0d997a9eeea |
|
30-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reserve RTP/RTCP modules in SetSSRC.
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
|
| 994d0b7229a18b255d81979c2bedaf8ecfae9bd7 |
|
27-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor Call-based tests.
Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.
BUG=3035
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 6ae48c660934784b4df56ab1ac99402ce3745e9f |
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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/full_stack.cc
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| 1566ee289364fdac5aa9dcc62db3070033208ad1 |
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23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
This reverts commit r6230 to re-land r6229.
ViECapturer::SwapFrame now resets timestamps.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 2cdd433edfec8d02c5f49fc634a8a07fc7e792ca |
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23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Remove VideoSendStreamInput::PutFrame."
This reverts r6229.
Test WebRtcVideoChannel2BaseTest.MuteStream fails after r6229.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6230 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| f3085e43ab7e0f6cb8c89cb02ed9e5694aba2e96 |
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23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove VideoSendStreamInput::PutFrame.
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| caba2d2a370cb6b5e67c881ecfa57fdac7411de8 |
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14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add DeliveryStatus enum to DeliverPacket().
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 023b101f4e0bf3c3be5562140cc99fbbdc36371c |
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13-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move gflags usage to video_loopback.
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.
R=mflodman@webrtc.org
BUG=3113
Review URL: https://webrtc-codereview.appspot.com/12379006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| de1429e9ad9a3a207ca191e1d748aa7271066860 |
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28-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add thread annotations to Call API.
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| a5c8d2c9b39a2d20fead2147e60ed0cd6d62019c |
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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/full_stack.cc
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| f577ae9eac9822380ea6f0fb953cf383d0ec5374 |
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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/full_stack.cc
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| 95153cc4cd9664afb04c5b45c8b49ad0e226108f |
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12-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove platform-specific code from new-API tests.
We've had problems that seem to manifest in run_tests.mm getting stuck
on exit. For our automated test targets only full_stack.cc was making
use of the platform-specific renderers provided by webrtc_test_common
and since no one currently monitors these the use case is hypothetical.
Readding platform-specific renderers to video_loopback is tracked with
issue 3039, though as far as I'm aware no one's currently using the
video_loopback target.
BUG=2987
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5686 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 724947b8efa44d15d699b471020005450590f5b6 |
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11-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SwapFrame() to VideoSendStreamInput.
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| b613b5ab2b03041942f04fd892e2ad5a4f9de027 |
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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/full_stack.cc
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| 2c46f8d854c1fc3e10f8151ee5109923287aee8b |
|
21-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename DestroyStream methods to include Video.
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 27326b6a42e8dc2431390aa016cbebfa2151aa05 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename newapi::Transport::SendRTP()->SendRtp().
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 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/full_stack.cc
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| 5a63655ab0de18bd2fa376ba4774eab3f3bc9fb2 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename Call::Create{Receive,Send}Stream().
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/full_stack.cc
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| 16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
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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/full_stack.cc
|