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}
On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.
This new method will be used in
https://codereview.webrtc.org/1422963003/
BUG=webrtc:4993
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1414693006 .
Cr-Commit-Position: refs/heads/master@{#10576}
Reason for revert:
Breaks bot.
Original issue's description:
> Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
>
> Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
>
> Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
>
> BUG=webrtc:5144, chromium:500602
>
> Committed: https://crrev.com/81c5c7f8157f767747bd97419eb0a589207354cf
> Cr-Commit-Position: refs/heads/master@{#10504}
TBR=stefan@webrtc.org,mflodman@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1423493005
Cr-Commit-Position: refs/heads/master@{#10508}
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1427253002
Cr-Commit-Position: refs/heads/master@{#10504}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
When estimating that we can send more than the codec max bitrate (under
the assumption that codec max bitrate is good enough for the quality),
we should use additional bitrate so that we can maintain good quality.
Global bitrate caps should still be enforced through bitrate caps (b=AS)
and not codec max bitrates.
BUG=webrtc:5102
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1428473002
Cr-Commit-Position: refs/heads/master@{#10457}
Encoders need to be externally provided. To use software encoders they
need to be created and registered from the outside.
BUG=webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1394823002 .
Cr-Commit-Position: refs/heads/master@{#10283}
In video_sender.cc, properly read the number of temporal layers for VP9 too.
Also, some cleanup in video_loopback.cc and video_quality_test.h.
Review URL: https://codereview.webrtc.org/1351693005
Cr-Commit-Position: refs/heads/master@{#10201}
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}
Fixes code formatting and uses size_t properly. Also makes use of
IsNewerTimestamp instead of a simple > check, which should fix an
edge-case bug.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1358863002
Cr-Commit-Position: refs/heads/master@{#10094}
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}
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.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
Handling the case when encoder drops only the higher layer.
Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers).
Review URL: https://codereview.webrtc.org/1287643002
Cr-Commit-Position: refs/heads/master@{#9883}
Especially the VP9 codec currently may overshoot bitrate target at sudden picture changes, resulting in frames over 800 packets.
This limit should be reduced again once the codec behaves.
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1266353003
Cr-Commit-Position: refs/heads/master@{#9675}
Enforces previous kProtectionKeyOnLoss as the permanent method which was
the only one used in use. This simplifies SetVideoProtection and
transition over to SetReceiverRobustnessMode.
BUG=webrtc:1596
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1244753002
Cr-Commit-Position: refs/heads/master@{#9641}
These payload types aren't directly connected to any payload type, and
the payload type still has to be negotiated externally. As such these
constants are just a source of confusion.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1215603003
Cr-Commit-Position: refs/heads/master@{#9546}
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}
This is just https://webrtc-codereview.appspot.com/53629004/
Remove a constructor of VCMJitterBuffer.
Remove unnecessary factory use
Comment Fix
Move frame incoming simulation to the clock
DCHECK typo fix
Coding Style Fix
Rephrased some comments, and removed some virtual for override function.
Coding Style Fix
Coding Style Fix
Add a unittest for VCMReceiver::FrameForDecoding. Mainly test the time control algorithm.
BUG=
TBR=holmer@chromium.org
Review URL: https://codereview.webrtc.org/1173253008.
Cr-Commit-Position: refs/heads/master@{#9470}
timestamp_ is only used in GenerateFrame() and its old value is
discarded. So it just needs to be a local variable in GenerateFrame().
As a result, we can remove the start_timestamp parameter from the
constructor and Init().
Also mark the GeneratePacket() method private because it is only used
internally.
R=stefan@webrtc.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/50149004
Cr-Commit-Position: refs/heads/master@{#9386}
