28 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
sergeyu
2cb155aa8a Remove deprected functions from EncodedImageCallback and RtpRtcp
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.

BUG=chromium:621691

Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Original-Commit-Position: refs/heads/master@{#14923}
Cr-Commit-Position: refs/heads/master@{#14935}
2016-11-04 18:39:37 +00:00
kjellander
91b957d3e4 Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
Reason for revert:
Still breaks internal downstream project.
Sergey: Please update internal project before relanding this.

Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
> Cr-Commit-Position: refs/heads/master@{#14923}

TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.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/2479643002
Cr-Commit-Position: refs/heads/master@{#14925}
2016-11-03 18:53:50 +00:00
sergeyu
c681250aaa Remove deprected functions from EncodedImageCallback and RtpRtcp
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.

BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14923}
2016-11-03 18:06:42 +00:00
ehmaldonado
43a9dc0f93 Revert of move deprected functions from EncodedImageCallback and RtpRtcp (patchset #1 id:1 of https://codereview.webrtc.org/2467373003/ )
Reason for revert:
Made a mistake while reverting.

Original issue's description:
> Reland of move deprected functions from EncodedImageCallback and RtpRtcp (patchset #2 id:240001 of https://codereview.webrtc.org/2474433008/ )
>
> Reason for revert:
> Breaks everything
>
> Original issue's description:
> > Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
> >
> > Reason for revert:
> > This might be breaking projects downstream.
> >
> > Original issue's description:
> > > Remove deprected functions from EncodedImageCallback and RtpRtcp
> > >
> > > Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> > > These methods should no longer be used anywhere and it's safe to remove
> > > them.
> > >
> > > BUG=chromium:621691
> > >
> > > Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> > > Cr-Commit-Position: refs/heads/master@{#14902}
> >
> > TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=chromium:621691
> >
> > Committed: https://crrev.com/6c78307a21252c2dbd704f6d5e92a220fb722ed4
> > Cr-Commit-Position: refs/heads/master@{#14914}
>
> TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:621691
>
> Committed: https://crrev.com/a1d6cd64083a3c0173aeefe38425a56de8942745
> Cr-Commit-Position: refs/heads/master@{#14915}

TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.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/2477773002
Cr-Commit-Position: refs/heads/master@{#14916}
2016-11-03 14:52:42 +00:00
ehmaldonado
a1d6cd6408 Reland of move deprected functions from EncodedImageCallback and RtpRtcp (patchset #2 id:240001 of https://codereview.webrtc.org/2474433008/ )
Reason for revert:
Breaks everything

Original issue's description:
> Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
>
> Reason for revert:
> This might be breaking projects downstream.
>
> Original issue's description:
> > Remove deprected functions from EncodedImageCallback and RtpRtcp
> >
> > Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> > These methods should no longer be used anywhere and it's safe to remove
> > them.
> >
> > BUG=chromium:621691
> >
> > Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> > Cr-Commit-Position: refs/heads/master@{#14902}
>
> TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:621691
>
> Committed: https://crrev.com/6c78307a21252c2dbd704f6d5e92a220fb722ed4
> Cr-Commit-Position: refs/heads/master@{#14914}

TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.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/2467373003
Cr-Commit-Position: refs/heads/master@{#14915}
2016-11-03 14:39:04 +00:00
ehmaldonado
6c78307a21 Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
Reason for revert:
This might be breaking projects downstream.

Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> Cr-Commit-Position: refs/heads/master@{#14902}

TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.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/2474433008
Cr-Commit-Position: refs/heads/master@{#14914}
2016-11-03 14:33:25 +00:00
sergeyu
fa56584271 Remove deprected functions from EncodedImageCallback and RtpRtcp
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.

BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14902}
2016-11-02 20:14:24 +00:00
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
Henrik Kjellander
2557b86e76 modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.

To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).

Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test

BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417283007 .

Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 21:00:33 +00:00
kjellander
6f8ce060a2 common_video: rename interface -> include
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.

BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc

Review URL: https://codereview.webrtc.org/1418913006

Cr-Commit-Position: refs/heads/master@{#10659}
2015-11-16 21:52:31 +00:00
Peter Boström
69ccb33131 Remove redudant encoder rate calls.
Moves EncoderParameters update checks into GenericEncoder before calling
SetRates/SetChannelParameters as applicable.

Also removes CodecConfigParameters as a bonus.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1426953003 .

Cr-Commit-Position: refs/heads/master@{#10452}
2015-10-29 15:30:29 +00:00
Peter Boström
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
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
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
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
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
sprang@webrtc.org
346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers.
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.

BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
sprang@webrtc.org
8b8819262f Improve VideoSendStreamTest::MaxPacketSize
This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/, but was reverted because of flakiness. This new issue will correct that.

Patch Set 1 contains the code that was submitted in 4849004.

BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 10:05:17 +00:00
andrew@webrtc.org
797522f9f2 Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
> Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
> 
> BUG=2428
> R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/4849004

It caused a failure in video_engine_tests on the Linux Tsan bot.

TBR=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5240 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 17:42:32 +00:00
sprang@webrtc.org
7104fc1906 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 16:15:11 +00:00