132 Commits

Author SHA1 Message Date
hbos
228fb0e1ec H264: Don't disable compiler warning by using NO_THREAD_SAFETY_ANALYSIS.
The LockManagerOperation global function uses void** which does not work
well with the thread_annotations.h macros and produce compiler warnings
on clang (chromium default). Workarounds to this is hacky and unhelpful
so we disable the analysis for this function, which isn't helpful in
this case anyway.

webrtc_h264_config is no longer needed and is removed.

BUG=8090, 8119

Review-Url: https://codereview.webrtc.org/3000263002
Cr-Commit-Position: refs/heads/master@{#19425}
2017-08-21 12:17:17 +00:00
sprang
ba050a6d6d Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
Reason for revert:
Create reland CL to add fix to.

Original issue's description:
> Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
>
> Reason for revert:
> Speculative revet for breaking remoting_unittests in fyi bots.
> https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester
>
> Original issue's description:
> > Add a flags field to video timing extension.
> >
> > The rtp header extension for video timing shuold have an additional
> > field for signaling metadata, such as what triggered the extension for
> > this particular frame. This will allow separating frames select because
> > of outlier sizes from regular frames, for more accurate stats.
> >
> > This implementation is backwards compatible in that it can read video
> > timing extensions without the new flag field, but it always sends with
> > it included.
> >
> > BUG=webrtc:7594
> >
> > Review-Url: https://codereview.webrtc.org/3000753002
> > Cr-Commit-Position: refs/heads/master@{#19353}
> > Committed: cf5d485e14
>
> TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/2995953002
> Cr-Commit-Position: refs/heads/master@{#19360}
> Committed: f0f7378b05

TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,emircan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2996153002
Cr-Commit-Position: refs/heads/master@{#19405}
2017-08-18 09:51:12 +00:00
emircan
f0f7378b05 Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
Reason for revert:
Speculative revet for breaking remoting_unittests in fyi bots.
https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester

Original issue's description:
> Add a flags field to video timing extension.
>
> The rtp header extension for video timing shuold have an additional
> field for signaling metadata, such as what triggered the extension for
> this particular frame. This will allow separating frames select because
> of outlier sizes from regular frames, for more accurate stats.
>
> This implementation is backwards compatible in that it can read video
> timing extensions without the new flag field, but it always sends with
> it included.
>
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/3000753002
> Cr-Commit-Position: refs/heads/master@{#19353}
> Committed: cf5d485e14

TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2995953002
Cr-Commit-Position: refs/heads/master@{#19360}
2017-08-15 19:31:23 +00:00
sprang
cf5d485e14 Add a flags field to video timing extension.
The rtp header extension for video timing shuold have an additional
field for signaling metadata, such as what triggered the extension for
this particular frame. This will allow separating frames select because
of outlier sizes from regular frames, for more accurate stats.

This implementation is backwards compatible in that it can read video
timing extensions without the new flag field, but it always sends with
it included.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/3000753002
Cr-Commit-Position: refs/heads/master@{#19353}
2017-08-15 12:33:27 +00:00
brandtr
669ea1917e Rename WEBRTC_VIDEOPROCESSOR_H264_TESTS define to WEBRTC_USE_H264.
This is the name used in other parts of the code.

BUG=none

Review-Url: https://codereview.webrtc.org/2996463003
Cr-Commit-Position: refs/heads/master@{#19253}
2017-08-07 10:35:13 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
This reverts commit c8fa692ec44fd6ba4fa3d085ac3161a262fc18c5.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2964773002 .
Cr-Commit-Position: refs/heads/master@{#18872}
2017-07-01 14:42:25 +00:00
kjellander
c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00
philipel
83c97da593 Only append SPS/PPS to bitstream if supplied out of band.
BUG=chromium:721597

Review-Url: https://codereview.webrtc.org/2945853002
Cr-Commit-Position: refs/heads/master@{#18701}
2017-06-21 14:22:40 +00:00
ilnik
04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00
Magnus Jedvert
72dbe2a211 Revert "Revert "Update video_coding/codecs to new VideoFrameBuffer interface""
This reverts commit 88f94fa36aa61f7904d30251205c544ada2c4301.

Chromium code has been updated.

Original change's description:
> Revert "Update video_coding/codecs to new VideoFrameBuffer interface"
> 
> This reverts commit 20ebf4ede803cd4f628ef9378700f60b72f2eab0.
> 
> Reason for revert:
> 
> Suspect of breaking FYI bots.
> See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/9036 and others.
> 
> Sample logs:
> Backtrace:
> [5024:1036:0607/173649.857:FATAL:webrtc_video_frame_adapter.cc(98)] Check failed: false. 
> Backtrace:
> 	base::debug::StackTrace::StackTrace [0x02D04A37+55]
> 	base::debug::StackTrace::StackTrace [0x02CCBB8A+10]
> 	content::WebRtcVideoFrameAdapter::NativeToI420Buffer [0x0508AD71+305]
> 	webrtc::VideoFrameBuffer::ToI420 [0x0230BF67+39]
> 	webrtc::H264EncoderImpl::Encode [0x057E8D0B+267]
> 	webrtc::VCMGenericEncoder::Encode [0x057E0E34+333]
> 	webrtc::vcm::VideoSender::AddVideoFrame [0x057DED9B+796]
> 	webrtc::ViEEncoder::EncodeVideoFrame [0x057C00F6+884]
> 	webrtc::ViEEncoder::EncodeTask::Run [0x057C12D7+215]
> 	rtc::TaskQueue::PostTask [0x03EE5CFB+194]
> 	base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDCAA5+31]
> 	base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDEE86+22]
> 	base::debug::TaskAnnotator::RunTask [0x02D08289+409]
> 	base::MessageLoop::RunTask [0x02C8CEC1+1233]
> 	base::MessageLoop::DoWork [0x02C8C1AD+765]
> 	base::MessagePumpDefault::Run [0x02D0A20B+219]
> 	base::MessageLoop::Run [0x02C8C9DB+107]
> 	base::RunLoop::Run [0x02C89583+147]
> 	base::Thread::Run [0x02CBEFCD+173]
> 	base::Thread::ThreadMain [0x02CBFADE+622]
> 	base::PlatformThread::Sleep [0x02C9E1A2+290]
> 	BaseThreadInitThunk [0x75C3338A+18]
> 	RtlInitializeExceptionChain [0x773A9902+99]
> 	RtlInitializeExceptionChain [0x773A98D5+54]
> 
> Original change's description:
> > Update video_coding/codecs to new VideoFrameBuffer interface
> > 
> > This is a follow-up cleanup for CL
> > https://codereview.webrtc.org/2847383002/.
> > 
> > Bug: webrtc:7632
> > Change-Id: I47861d779968f2fee94db9c017102a8e87e67fb7
> > Reviewed-on: https://chromium-review.googlesource.com/524163
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#18477}
> 
> TBR=magjed@webrtc.org,nisse@webrtc.org,brandtr@webrtc.org
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7632
> 
> Change-Id: I3b73fc7d16ff19ceba196e964dcb36a36510912c
> Reviewed-on: https://chromium-review.googlesource.com/527793
> Reviewed-by: Guido Urdaneta <guidou@chromium.org>
> Commit-Queue: Guido Urdaneta <guidou@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#18489}

TBR=tterriberry@mozilla.com,mflodman@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,guidou@chromium.org,nisse@webrtc.org,brandtr@webrtc.org,webrtc-reviews@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Presubmit: true
Bug: webrtc:7632

Change-Id: I0962a704e8a9939d4364ce9069c863c9951654c9
Reviewed-on: https://chromium-review.googlesource.com/530684
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18527}
2017-06-10 20:12:17 +00:00
Guido Urdaneta
88f94fa36a Revert "Update video_coding/codecs to new VideoFrameBuffer interface"
This reverts commit 20ebf4ede803cd4f628ef9378700f60b72f2eab0.

Reason for revert:

Suspect of breaking FYI bots.
See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/9036 and others.

Sample logs:
Backtrace:
[5024:1036:0607/173649.857:FATAL:webrtc_video_frame_adapter.cc(98)] Check failed: false. 
Backtrace:
	base::debug::StackTrace::StackTrace [0x02D04A37+55]
	base::debug::StackTrace::StackTrace [0x02CCBB8A+10]
	content::WebRtcVideoFrameAdapter::NativeToI420Buffer [0x0508AD71+305]
	webrtc::VideoFrameBuffer::ToI420 [0x0230BF67+39]
	webrtc::H264EncoderImpl::Encode [0x057E8D0B+267]
	webrtc::VCMGenericEncoder::Encode [0x057E0E34+333]
	webrtc::vcm::VideoSender::AddVideoFrame [0x057DED9B+796]
	webrtc::ViEEncoder::EncodeVideoFrame [0x057C00F6+884]
	webrtc::ViEEncoder::EncodeTask::Run [0x057C12D7+215]
	rtc::TaskQueue::PostTask [0x03EE5CFB+194]
	base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDCAA5+31]
	base::internal::Invoker<base::internal::BindState<enum extensions::`anonymous namespace'::VerificationResult (__cdecl*)(std::unique_ptr<extensions::NetworkingCastPrivateDelegate::Credentials,std::default_delete<extensions::NetworkingCastPrivateDelegate::C [0x02DDEE86+22]
	base::debug::TaskAnnotator::RunTask [0x02D08289+409]
	base::MessageLoop::RunTask [0x02C8CEC1+1233]
	base::MessageLoop::DoWork [0x02C8C1AD+765]
	base::MessagePumpDefault::Run [0x02D0A20B+219]
	base::MessageLoop::Run [0x02C8C9DB+107]
	base::RunLoop::Run [0x02C89583+147]
	base::Thread::Run [0x02CBEFCD+173]
	base::Thread::ThreadMain [0x02CBFADE+622]
	base::PlatformThread::Sleep [0x02C9E1A2+290]
	BaseThreadInitThunk [0x75C3338A+18]
	RtlInitializeExceptionChain [0x773A9902+99]
	RtlInitializeExceptionChain [0x773A98D5+54]

Original change's description:
> Update video_coding/codecs to new VideoFrameBuffer interface
> 
> This is a follow-up cleanup for CL
> https://codereview.webrtc.org/2847383002/.
> 
> Bug: webrtc:7632
> Change-Id: I47861d779968f2fee94db9c017102a8e87e67fb7
> Reviewed-on: https://chromium-review.googlesource.com/524163
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18477}

TBR=magjed@webrtc.org,nisse@webrtc.org,brandtr@webrtc.org
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7632

Change-Id: I3b73fc7d16ff19ceba196e964dcb36a36510912c
Reviewed-on: https://chromium-review.googlesource.com/527793
Reviewed-by: Guido Urdaneta <guidou@chromium.org>
Commit-Queue: Guido Urdaneta <guidou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#18489}
2017-06-08 08:33:52 +00:00
Magnus Jedvert
20ebf4ede8 Update video_coding/codecs to new VideoFrameBuffer interface
This is a follow-up cleanup for CL
https://codereview.webrtc.org/2847383002/.

Bug: webrtc:7632
Change-Id: I47861d779968f2fee94db9c017102a8e87e67fb7
Reviewed-on: https://chromium-review.googlesource.com/524163
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18477}
2017-06-07 15:58:13 +00:00
brandtr
080830c513 Don't re-randomize picture_id/tl0_pic_idx when re-initializing internal encoders.
TESTED=video_loopback and AppRTCMobile with forced encoder reinits every 30 frames.
BUG=webrtc:7475

Review-Url: https://codereview.webrtc.org/2833493003
Cr-Commit-Position: refs/heads/master@{#17984}
2017-05-03 10:25:53 +00:00
nisse
eb44b39a21 Delete RawVideoType enum, use the VideoType enum instead.
BUG=webrtc:7385

Review-Url: https://codereview.webrtc.org/2765243002
Cr-Commit-Position: refs/heads/master@{#17930}
2017-04-28 14:18:05 +00:00
ilnik
00d802b6ee Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
Reason for revert:
Fix failing bots.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816493002
Cr-Commit-Position: refs/heads/master@{#17658}
2017-04-11 17:34:31 +00:00
ilnik
27c46e2872 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 13:20:05 +00:00
ilnik
774f6b4b96 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with appropriate changes to API to not break depending projects.

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420

Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
2017-04-11 13:12:37 +00:00
ilnik
29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
Reason for revert:
Relanded by mistake.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97f

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420

Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
2017-04-11 11:49:07 +00:00
ilnik
4fa0c4f97f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with fixes which break API

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420

Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
2017-04-11 11:01:43 +00:00
ilnik
5721866808 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
Reason for revert:
Breaks dependent projects.

Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeae

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@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:7420

Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
2017-04-11 10:59:43 +00:00
ilnik
64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header.
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.

Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
2017-04-11 08:46:04 +00:00
sakal
b8102e0634 Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ )
Reason for revert:
The issue is now hopefully fixed.

Original issue's description:
> Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ )
>
> Reason for revert:
> Let's revert this while we investigate a problem in H264 bitstream parser.
>
> Original issue's description:
> > Add QP for FFmpeg H264 decoder.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133007
> > Cr-Commit-Position: refs/heads/master@{#16942}
> > Committed: 879f4f6c31
>
> TBR=sprang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6541, chromium:697795
>
> Review-Url: https://codereview.webrtc.org/2726973003
> Cr-Commit-Position: refs/heads/master@{#16974}
> Committed: 4c6df8893e

TBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6541, chromium:697795

Review-Url: https://codereview.webrtc.org/2735733002
Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 11:52:55 +00:00
sakal
4c6df8893e Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ )
Reason for revert:
Let's revert this while we investigate a problem in H264 bitstream parser.

Original issue's description:
> Add QP for FFmpeg H264 decoder.
>
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2649133007
> Cr-Commit-Position: refs/heads/master@{#16942}
> Committed: 879f4f6c31

TBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6541, chromium:697795

Review-Url: https://codereview.webrtc.org/2726973003
Cr-Commit-Position: refs/heads/master@{#16974}
2017-03-02 15:22:26 +00:00
sakal
879f4f6c31 Add QP for FFmpeg H264 decoder.
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133007
Cr-Commit-Position: refs/heads/master@{#16942}
2017-03-01 14:19:09 +00:00
sakal
a2508c1e0f Fix reporting QP in FFmpeg H264 encoder.
The QP was previously written after calling OnEncodedImage. This was a
bug.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2718353003
Cr-Commit-Position: refs/heads/master@{#16911}
2017-02-28 15:57:00 +00:00
sprang
10418acb97 Remove backwards compatibilty path for OpenH264 v1.4
Deps have rolled to 1.6, and since no one noticed that the old code path
was broken and wouldn't even compile, I assume no one is using it.
I therefore deem it time to clean away all these nasty ifdefs.

("const kNalHeaderSizeAllocation = 50;" doesn't declare a type)

BUG=chromium:614970

Review-Url: https://codereview.webrtc.org/2622233002
Cr-Commit-Position: refs/heads/master@{#16008}
2017-01-11 13:51:56 +00:00
nisse
af916899cc Move VideoFrame and related declarations to webrtc/api/video.
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.

BUG=webrtc:5880

Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
2017-01-10 15:44:26 +00:00
nisse
df2ceb88a8 Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
Reason for revert:
Fixing perf tests.

Original issue's description:
> Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
>
> Reason for revert:
> Crashes perf tests, e.g.,
>
> ./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
>
> dies with an assert related to rtc::Optional.
>
> Original issue's description:
> > Delete VideoFrame default constructor, and IsZeroSize method.
> >
> > This ensures that the video_frame_buffer method never can return a
> > null pointer.
> >
> > BUG=webrtc:6591
> >
> > Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> > Cr-Commit-Position: refs/heads/master@{#15574}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6591
>
> Committed: https://crrev.com/0989fbcad2ca4eb5805a77e8ebfefd3af06ade23
> Cr-Commit-Position: refs/heads/master@{#15597}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574183002
Cr-Commit-Position: refs/heads/master@{#15633}
2016-12-15 14:30:00 +00:00
hta
41286496cb Move all codec specific definitions from modules_include
This CL moves all codec specific definitions into their own
header files in the respective codec directory, and replaces
it with an include in the top level directory.

This is to facilitate getting the code out of the header files.

No behavioral changes are expected.

BUG=webrtc:6842

Review-Url: https://codereview.webrtc.org/2555993003
Cr-Commit-Position: refs/heads/master@{#15623}
2016-12-15 08:54:15 +00:00
nisse
0989fbcad2 Revert of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #5 id:80001 of https://codereview.webrtc.org/2541863002/ )
Reason for revert:
Crashes perf tests, e.g.,

./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'

dies with an assert related to rtc::Optional.

Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
2016-12-14 10:06:49 +00:00
nisse
bfcf561923 Delete VideoFrame default constructor, and IsZeroSize method.
This ensures that the video_frame_buffer method never can return a
null pointer.

BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
2016-12-13 14:08:39 +00:00
hta
9aa96889a3 Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
Reason for revert:
Fixed timeouts in slow tests

Original issue's description:
> Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
>
> Reason for revert:
> Failures on the Linux Memcheck bot
>
> Original issue's description:
> > This approach passes packetization mode to the encoder as part of
> > a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
> >
> > BUG=600254
> >
> > Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> > Cr-Commit-Position: refs/heads/master@{#15437}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=600254
>
> Committed: https://crrev.com/243a0a7a7fd6b5da1e32df31f1bfbb6a68dc09f3
> Cr-Commit-Position: refs/heads/master@{#15441}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558463002
Cr-Commit-Position: refs/heads/master@{#15445}
2016-12-06 13:36:13 +00:00
hta
243a0a7a7f Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
Reason for revert:
Failures on the Linux Memcheck bot

Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
2016-12-06 12:22:05 +00:00
hta
e59647b991 This approach passes packetization mode to the encoder as part of
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.

BUG=600254

Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
2016-12-06 10:22:54 +00:00
kthelgason
876222f77d Move usage of QualityScaler to ViEEncoder.
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
  encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
  having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
  but has a callback to ViEEncoder that it uses to express it's desire
  for lower resolution.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
2016-11-29 09:44:22 +00:00
kwiberg
352444fcac RTC_[D]CHECK_op: Remove superfluous casts
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.

It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
2016-11-28 23:59:03 +00:00
magjed
ceecea4559 Pass selected cricket::VideoCodec down to internal H264 encoder
Pass the selected cricket::VideoCodec to H264EncoderImpl::H264EncoderImpl. The cricket::VideoCodec contains relevant information for H264 about selected profile and packetization mode.

BUG=chromium:600254,webrtc:6402, webrtc:6337

Review-Url: https://codereview.webrtc.org/2474993002
Cr-Commit-Position: refs/heads/master@{#15270}
2016-11-28 15:20:26 +00:00
magjed
2d60e53ad5 H264 encoder: Include QP information in encoded images
Set the |qp_| field in EncodedImage before calling OnEncodedImage.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2499003002
Cr-Commit-Position: refs/heads/master@{#15199}
2016-11-22 16:42:14 +00:00
Henrik Kjellander
b4af3d673a Remove all references to GYP
Remove all .gyp and .gypi files.
Remove entries from OWNERS files for *.isolate, *.gyp, *.gypi
Remove unused scripts in webrtc/build.

BUG=webrtc:6323
R=henrika@webrtc.org, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/2509703002 .

Cr-Commit-Position: refs/heads/master@{#15107}
2016-11-16 19:11:38 +00:00
Erik Språng
08127a9449 Reland #2 of Issue 2434073003: Extract bitrate allocation ...
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:

1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.

Please review only the changes after patch set 1.

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
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2510583002 .

Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 15:41:45 +00:00
magjed
13ceeeadfc Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ )
Reason for revert:
Broke a lot of tests in chromium.webrtc browser_tests. See e.g. https://build.chromium.org/p/chromium.webrtc/builders/Mac%20Tester/builds/62228 and https://build.chromium.org/p/chromium.webrtc/builders/Win8%20Tester/builds/30102.
[ RUN      ] WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264/1
...
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\modules\rtp_rtcp\source\rtp_format_h264.cc, line 170
# last system error: 0
# Check failed: packetization_mode_ == kH264PacketizationMode1 (0 vs. 2)
#

Original issue's description:
> Implement H.264 packetization mode 0.
>
> This approach extends the H.264 specific information with
> a packetization mode enum.
>
> Status: Parameter is in code. No way to set it yet.
>
> Rebase of CL  2009213002
>
> BUG=600254
>
> Committed: https://crrev.com/3bba101f36483b8030a693dfbc93af736d1dba68
> Cr-Commit-Position: refs/heads/master@{#15032}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600254
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2500743002
Cr-Commit-Position: refs/heads/master@{#15050}
2016-11-12 16:54:50 +00:00
hta
3bba101f36 Implement H.264 packetization mode 0.
This approach extends the H.264 specific information with
a packetization mode enum.

Status: Parameter is in code. No way to set it yet.

Rebase of CL  2009213002

BUG=600254

Review-Url: https://codereview.webrtc.org/2337453002
Cr-Commit-Position: refs/heads/master@{#15032}
2016-11-11 05:50:05 +00:00
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
Kári Tristan Helgason
cbe7435288 Reland of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #1 id:1 of https://codereview.webrtc.org/2483273002/ )
Reason for revert:
Fix gyp build

Original issue's description:
> Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ )
>
> Reason for revert:
> Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes.
>
> Original issue's description:
> > Add a webrtc{en,de}coderfactory implementation for VideoToolbox
> >
> > This CL removes the coupling of the VideoToolbox h264 implementation
> > to the generic h264 code. The files have been moved into sdb/obj/Framework
> > and all dependency on them has been removed from the rest of WebRTC.
> > We now add it as an external encoder via a factory supplied to the
> > CreatePeerConnectionFactory call. This also brings the iOS implementation
> > closer to what we do on Android for MediaCodec.
> >
> > BUG=webrtc:6619
> >
> > Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4
> > Cr-Commit-Position: refs/heads/master@{#14953}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
> BUG=webrtc:6619
>
> Committed: https://crrev.com/d69ad84420d9c0e1c11450c352f6c92e7c9583f1
> Cr-Commit-Position: refs/heads/master@{#14985}

R=magjed@webrtc.org
TBR=kwiberg@webrtc.org, magjed@webrtc.org, stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6619

Review URL: https://codereview.webrtc.org/2487723004 .

Cr-Commit-Position: refs/heads/master@{#14992}
2016-11-09 09:43:38 +00:00
kwiberg
d69ad84420 Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ )
Reason for revert:
Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes.

Original issue's description:
> Add a webrtc{en,de}coderfactory implementation for VideoToolbox
>
> This CL removes the coupling of the VideoToolbox h264 implementation
> to the generic h264 code. The files have been moved into sdb/obj/Framework
> and all dependency on them has been removed from the rest of WebRTC.
> We now add it as an external encoder via a factory supplied to the
> CreatePeerConnectionFactory call. This also brings the iOS implementation
> closer to what we do on Android for MediaCodec.
>
> BUG=webrtc:6619
>
> Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4
> Cr-Commit-Position: refs/heads/master@{#14953}

TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
BUG=webrtc:6619

Review-Url: https://codereview.webrtc.org/2483273002
Cr-Commit-Position: refs/heads/master@{#14985}
2016-11-08 18:42:55 +00:00
kthelgason
6a5047dad3 Add a webrtc{en,de}coderfactory implementation for VideoToolbox
This CL removes the coupling of the VideoToolbox h264 implementation
to the generic h264 code. The files have been moved into sdb/obj/Framework
and all dependency on them has been removed from the rest of WebRTC.
We now add it as an external encoder via a factory supplied to the
CreatePeerConnectionFactory call. This also brings the iOS implementation
closer to what we do on Android for MediaCodec.

BUG=webrtc:6619

Review-Url: https://codereview.webrtc.org/2463313002
Cr-Commit-Position: refs/heads/master@{#14953}
2016-11-07 15:26:05 +00:00