49 Commits

Author SHA1 Message Date
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
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
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
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
palmkvist
e75f204b06 Expose Ivf logging through the native API
BUG=webrtc:6300

Review-Url: https://codereview.webrtc.org/2303273002
Cr-Commit-Position: refs/heads/master@{#14419}
2016-09-28 13:19:53 +00:00
perkj
26091b1118 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/

- Add task queue to Call with the intent of replacing the use of one of the process threads.

- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

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

Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
2016-09-01 08:17:43 +00:00
perkj
275afc5d2e Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2253563002
Cr-Commit-Position: refs/heads/master@{#14012}
2016-09-01 07:21:18 +00:00
perkj
8eb37a39e7 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
Reason for revert:
Failed on Win 10 Chrome FYI.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio

#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#

WebRtcBrowserTest

#

Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}

TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
2016-08-16 09:40:59 +00:00
perkj
cc168360f4 - Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
2016-08-16 07:38:51 +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
Erik Språng
ad113e50d2 Fix bug in calculation of averge queue time in paced sender.
Also work around a flaw in fake encoder which caused bogus perf
regression in rampup tests.

BUG=560434
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1474533006 .

Cr-Commit-Position: refs/heads/master@{#10811}
2015-11-26 15:26:25 +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
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
Peter Boström
5d0379da2c Remove kSkipFrame usage.
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.

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

Review URL: https://codereview.webrtc.org/1369923005 .

Cr-Commit-Position: refs/heads/master@{#10181}
2015-10-06 12:05:03 +00:00
asapersson
d89920b74a Add resolution and fps stats to histograms:
- "WebRTC.Video.InputWidthInPixels"
- "WebRTC.Video.InputHeightInPixels"
- "WebRTC.Video.SentWidthInPixels"
- "WebRTC.Video.SentHeightInPixels"
- "WebRTC.Video.ReceivedWidthInPixels"
- "WebRTC.Video.ReceivedHeightInPixels"
- "WebRTC.Video.RenderFramesPerSecond"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1228393008

Cr-Commit-Position: refs/heads/master@{#9611}
2015-07-22 13:52:03 +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
glaznev@webrtc.org
e69220ca84 Fix the value of the first byte of nal unit
generated by fake H.264 encoder.

BUG=
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38849004

Cr-Commit-Position: refs/heads/master@{#8253}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8253 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 17:56:44 +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
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +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
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
sprang@webrtc.org
4070935f4f Implement and test EncodedImageCallback in new ViE API.
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-26 11:41:59 +00:00
stefan@webrtc.org
69969e2e2f Improve Call tests for RTX.
Also does some refactoring to reuse RtpRtcpObserver.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 12:32:15 +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