30 Commits

Author SHA1 Message Date
sprang
1369c83b42 Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ )
Reason for revert:
Seems to be causing flakiness in perf test:
FullStackTest.ScreenshareSlidesVP8_2TL_LossyNet

Original issue's description:
> Reland of Issue 2434073003: Extract bitrate allocation ...
>
> This is a reland of https://codereview.webrtc.org/2434073003/ including
> some fixes for failing test cases.
>
> Original description:
>
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
>   of how the target bitrate is distributed over spatial and temporal
>   layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
>   rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
>   SimulcastRateAllocator is extended to fully handle VP8 allocation.
>   This includes capturing TemporalLayer instances created by the
>   encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
>   factories for VP8. This allows allocation to happen fully outside of
>   the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/647bf43dcb2fd16fccf276bd94dc4400728bb405
> Cr-Commit-Position: refs/heads/master@{#15023}

TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2491393002
Cr-Commit-Position: refs/heads/master@{#15026}
2016-11-10 16:30:39 +00:00
sprang
647bf43dcb Reland of Issue 2434073003: Extract bitrate allocation ...
This is a reland of https://codereview.webrtc.org/2434073003/ including
some fixes for failing test cases.

Original description:

Extract bitrate allocation of spatial/temporal layers out of codec impl.

This CL makes a number of intervowen changes:

* Add BitrateAllocation struct, that contains a codec independent view
  of how the target bitrate is distributed over spatial and temporal
  layers.

* Adds the BitrateAllocator interface, which takes a bitrate and frame
  rate and produces a BitrateAllocation.

* A default (non layered) implementation is added, and
  SimulcastRateAllocator is extended to fully handle VP8 allocation.
  This includes capturing TemporalLayer instances created by the
  encoder.

* ViEEncoder now owns both the bitrate allocator and the temporal layer
  factories for VP8. This allows allocation to happen fully outside of
  the encoder implementation.

This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2488833004
Cr-Commit-Position: refs/heads/master@{#15023}
2016-11-10 14:46:28 +00:00
sprang
4bc98d4e1b Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
Reason for revert:
Breaks perf tests.

Original issue's description:
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
>   of how the target bitrate is distributed over spatial and temporal
>   layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
>   rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
>   SimulcastRateAllocator is extended to fully handle VP8 allocation.
>   This includes capturing TemporalLayer instances created by the
>   encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
>   factories for VP8. This allows allocation to happen fully outside of
>   the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/8f46c679d24a05b3f08e02c6d91ec9637f34e24f
> Cr-Commit-Position: refs/heads/master@{#14998}

TBR=stefan@webrtc.org,perkj@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2489843002
Cr-Commit-Position: refs/heads/master@{#15001}
2016-11-09 14:14:56 +00:00
sprang
8f46c679d2 Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:

* Add BitrateAllocation struct, that contains a codec independent view
  of how the target bitrate is distributed over spatial and temporal
  layers.

* Adds the BitrateAllocator interface, which takes a bitrate and frame
  rate and produces a BitrateAllocation.

* A default (non layered) implementation is added, and
  SimulcastRateAllocator is extended to fully handle VP8 allocation.
  This includes capturing TemporalLayer instances created by the
  encoder.

* ViEEncoder now owns both the bitrate allocator and the temporal layer
  factories for VP8. This allows allocation to happen fully outside of
  the encoder implementation.

This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2434073003
Cr-Commit-Position: refs/heads/master@{#14998}
2016-11-09 13:09:12 +00:00
perkj
803d97f159 Let ViEEncoder express resolution requests as Sinkwants.
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.

To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.

BUG=webrtc:5687
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
2016-11-01 18:45:54 +00:00
Sergey Ulanov
525df3ffd1 Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Original-Commit-Position: refs/heads/master@{#13615}
Cr-Commit-Position: refs/heads/master@{#13617}
2016-08-03 00:46:47 +00:00
sergeyu
51db4dd1bd Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ )
Reason for revert:
broke browser_tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
2016-08-03 00:33:47 +00:00
Sergey Ulanov
4c7f4cd2ef Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Commit-Position: refs/heads/master@{#13615}
2016-08-02 22:14:51 +00:00
sergeyu
ac4dc2cefe Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ )
Reason for revert:
broke internal tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
2016-08-02 21:33:21 +00:00
Sergey Ulanov
ad34dbe934 Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Cr-Commit-Position: refs/heads/master@{#13613}
2016-08-02 20:44:25 +00:00
Peter Boström
b7d9a97ce4 Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.

BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1406903002 .

Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18 15:01:23 +00:00
Peter Boström
7623ce4aeb Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.

Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}

BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1510183002 .

Cr-Commit-Position: refs/heads/master@{#10948}
2015-12-09 11:13:40 +00:00
kjellander
8237abf563 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.

Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}

TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695

Review URL: https://codereview.webrtc.org/1507903005

Cr-Commit-Position: refs/heads/master@{#10937}
2015-12-08 15:12:11 +00:00
Peter Boström
03ef053202 Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1506773002 .

Cr-Commit-Position: refs/heads/master@{#10926}
2015-12-08 08:09:07 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
pbos
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
Miguel Casas-Sanchez
4765070b8d 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}
2015-05-30 00:21:56 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: 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
2015-03-04 13:04:54 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream.
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
asapersson@webrtc.org
049e4ece30 Change default values for CpuOveruseOptions.
Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85).

Moved DelayedEncoder to fake_encoder.h and added configuration for the delay.

R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7722 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 10:19:46 +00:00
pbos@webrtc.org
ab990ae43a 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
2014-09-17 09:02:25 +00:00
henrikg@webrtc.org
307d3dbdee 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
2014-09-11 09:48:30 +00:00
pbos@webrtc.org
b420191743 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
2014-09-09 10:40:56 +00:00
stefan@webrtc.org
79c3359e67 Add end-to-end H.264 packetization test.
Also correctly wires up H.264 packetization in the new Call api.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06 09:24:53 +00:00
pbos@webrtc.org
de1429e9ad 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
2014-04-28 13:00:21 +00:00
pbos@webrtc.org
f577ae9eac 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
2014-03-19 08:43:57 +00:00
pbos@webrtc.org
3349ae0cdc Implement minimum transmit bitrate.
Utilizing minimum transmission bitrate prevents low remote bitrate
estimates (bitrate estimation dips) when encoding non-complex content
such as screenshare of a static image even though there's nothing wrong
with the link.

Requires pacing to be enabled for now, pending issue 3036.

BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 12:52:27 +00:00
pbos@webrtc.org
16e03b7bd8 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
2013-10-28 16:32:01 +00:00