237 Commits

Author SHA1 Message Date
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
Erik Språng
737336d37a Add NACK rate throttling for audio channels.
Not really used for audio today (already in place for video), but should
still function anyway.

BUG=
R=henrika@webrtc.org, minyue@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2181383002 .

Cr-Commit-Position: refs/heads/master@{#13571}
2016-07-29 10:59:49 +00:00
Stefan Holmer
be40296cce Fix bug where a connection switch causes BWE to be set to zero.
BUG=webrtc:6076
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2125523004 .

Cr-Commit-Position: refs/heads/master@{#13414}
2016-07-08 14:16:52 +00:00
sprang
9c0b551425 Fix bug where min transmit bitrate wasn't working
A recent refactoring (r13192) introduced a bug where the min transmit
config wasn't being respected. Specifically, if a VideoSendStream was
created without it and the reconfigured, the min transmit bitrate would
not take effect. Probably the other way around as well.

BUG=webrtc::5687

Review-Url: https://codereview.webrtc.org/2106183002
Cr-Commit-Position: refs/heads/master@{#13390}
2016-07-06 07:54:35 +00:00
perkj
57c21f9b44 Remove ViEEncoder::Pause / Start
This cl change so that VideoSendStream::Start adds the stream as a BitrateObserver and VideoSendStream::Stop removes the stream as observer.

That also means that start will trigger a VideoEncoder::SetRate call with the most recent bitrate estimate.
VideoSendStream::Stop will trigger a VideoEncoder::SetRate with bitrate  = 0.

BUG=webrtc:5687 b/28636240

Review-Url: https://codereview.webrtc.org/2070343002
Cr-Commit-Position: refs/heads/master@{#13192}
2016-06-17 14:27:23 +00:00
perkj
71ee44cc6d This cl:
1. It moves calculation of the needed padding to VideoSendStream instead of ViEEncoder and only does it once per send Stream instead of every time the network estimate changes.

2. The maximum amount of padding sent was prior to this cl calculated and updated based on network estimate changes. However, it can only change based on encoder configuration changes and if send streams are added or removed. This cl change the VideoSendStream/VieEncoder to notify the BitrateAllocator of changes to the needed padding bitrate and for BitrateAllocator to notify Call of these changes.

3. Fixed an issue in the SendPacer where it could send a padding packet before sending a real packet. This caused the test EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx to fail with these refactorings since the pacer suddenly could send a padding packet before the encoder had produced its first frame.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1993113003
Cr-Commit-Position: refs/heads/master@{#13149}
2016-06-15 07:47:58 +00:00
Peter Boström
0208322ee3 GN: Add video_engine_tests
Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 10:53:09 +00:00
mflodman
101f250a30 Implementing auto pausing of video streams.
This CL implements auto pausing video streams per stream with logic to
avoid toggling state too often.

Also re-enabling tests disabled for Mac, with the assumption the new
logic removes flakiness.

BUG=webrtc:5868,webrtc:5407
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2035383002 .

Cr-Commit-Position: refs/heads/master@{#13092}
2016-06-09 15:21:30 +00:00
stefan
8f4c77fea1 Always send RED headers if configured.
This shouldn't be needed, but because the receiver assumes RTX packets
contain RED if configured to receive them (due to an incompatibility
issue), we also have to make sure we send them for now.

BUG=webrtc:5675

Review-Url: https://codereview.webrtc.org/2033763002
Cr-Commit-Position: refs/heads/master@{#13024}
2016-06-03 07:16:54 +00:00
isheriff
6f8d686d35 Remove use of RtpHeaderExtension and clean up
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension

The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.

Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.

BUG= webrtc:5895

Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
2016-05-26 18:25:04 +00:00
Peter Boström
1299615838 Make sure WebRTC works without libvpx VP9 support.
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.

BUG=webrtc:5884
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1970343002 .

Cr-Commit-Position: refs/heads/master@{#12741}
2016-05-14 00:03:28 +00:00
philipel
696a802332 Re-enable Vp9FlexModeRefCount
Looks like this test was disable (https://codereview.webrtc.org/1556273002) but never re-enabled after the bug was fixed.

BUG=webrtc:5402

Review-Url: https://codereview.webrtc.org/1914893003
Cr-Commit-Position: refs/heads/master@{#12606}
2016-05-03 12:45:48 +00:00
nisse
ef8b61e110 Enable -Winconsistent-missing-override flag.
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.

NOPRESUBMIT=True
BUG=webrtc:3970

Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
2016-04-29 13:09:23 +00:00
philipel
c6957c7b73 De-flake VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly
BUG=webrtc:5382
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1922483002 .

Cr-Commit-Position: refs/heads/master@{#12547}
2016-04-28 13:52:59 +00:00
pbos
14fe708f3d Reland of Initialize/configure video encoders asychronously. (patchset #1 id:1 of https://codereview.webrtc.org/1821983002/ )
Reason for revert:
RTCVideoEncoder has been updated to not make assumptions on calling threads/post back to a worker thread. This should now be landable again.

Original issue's description:
> Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ )
>
> Reason for revert:
> Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
>
> Original issue's description:
> > Initialize/configure video encoders asychronously.
> >
> > Greatly speeds up setRemoteDescription() by moving encoder initialization
> > off the main worker thread, which is free to move onto gathering ICE
> > candidates and other tasks while InitEncode() is performed. It also
> > un-blocks PeerConnection GetStats() which is no longer blocked on
> > encoder initialization.
> >
> > BUG=webrtc:5410
> > R=stefan@webrtc.org
> >
> > Committed: fb647a67be
>
> R=stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:595274, chromium:595308, webrtc:5410
>
> Committed: https://crrev.com/81cbd924447d507559dbd6e6d1f9fe439fcf2716
> Cr-Commit-Position: refs/heads/master@{#12086}

TBR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410

Review URL: https://codereview.webrtc.org/1896413002

Cr-Commit-Position: refs/heads/master@{#12446}
2016-04-20 13:36:05 +00:00
kjellander
02b3d275a0 Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #1 id:1 of https://codereview.webrtc.org/1903193002/ )
Reason for revert:
A fix is being prepared downstream so this can now go in.

Original issue's description:
> Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
>
> Reason for revert:
> API changes broke downstream.
>
> Original issue's description:
> > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> > EncodedImageCallback can of course be cleaned up in the future.
> >
> > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
> >
> > BUG=webrtc::5687
> >
> > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> > Cr-Commit-Position: refs/heads/master@{#12436}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5687
>
> Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4
> Cr-Commit-Position: refs/heads/master@{#12441}

TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@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/1905583002

Cr-Commit-Position: refs/heads/master@{#12442}
2016-04-20 12:06:01 +00:00
kjellander
a261e61366 Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
Reason for revert:
API changes broke downstream.

Original issue's description:
> Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> EncodedImageCallback can of course be cleaned up in the future.
>
> This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
>
> BUG=webrtc::5687
>
> Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> Cr-Commit-Position: refs/heads/master@{#12436}

TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@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/1903193002

Cr-Commit-Position: refs/heads/master@{#12441}
2016-04-20 11:13:30 +00:00
perkj
f5d55aaecd Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
EncodedImageCallback can of course be cleaned up in the future.

This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.

BUG=webrtc::5687

Review URL: https://codereview.webrtc.org/1897233002

Cr-Commit-Position: refs/heads/master@{#12436}
2016-04-20 08:17:11 +00:00
pbos
a96b60b3a6 Move frame_callback.h to common_video/include.
BUG=webrtc:4243
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1902543002

Cr-Commit-Position: refs/heads/master@{#12419}
2016-04-19 04:12:57 +00:00
nisse
d30a1117f8 Change pre_encode_callback to get a const frame.
Used only by tests. Deleted the EndToEndTest.UsesFrameCallbacks, which
modified pixel data. Change callback from in EndToEndTest.GetStats to call SleepMs, rath than
modifying the timestamp.

BUG=

Review URL: https://codereview.webrtc.org/1891733002

Cr-Commit-Position: refs/heads/master@{#12406}
2016-04-18 12:15:33 +00:00
nisse
26acec4772 Delete method webrtc::VideoFrame::native_handle.
Instead, use the corresponding method on VideoFrameBuffer. In the process,
reduce code duplication in frame comparison functions used in
the test code.

Make FramesEqual use FrameBufsEqual. Make the latter support texture frames.

The cl also refactors VideoFrame::CopyFrame to use I420Buffer::Copy. This
has possibly undesired side effects of never reusing the frame buffer of
the destination frame, and producing a frame buffer which may use different
stride than the source frame.

BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1881953002

Cr-Commit-Position: refs/heads/master@{#12373}
2016-04-15 10:43:45 +00:00
Peter Boström
74f6e9ea23 Replace NULL with nullptr in webrtc/video.
BUG=
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/1855433002 .

Cr-Commit-Position: refs/heads/master@{#12218}
2016-04-04 15:56:22 +00:00
Peter Boström
81cbd92444 Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ )
Reason for revert:
Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.

Original issue's description:
> Initialize/configure video encoders asychronously.
>
> Greatly speeds up setRemoteDescription() by moving encoder initialization
> off the main worker thread, which is free to move onto gathering ICE
> candidates and other tasks while InitEncode() is performed. It also
> un-blocks PeerConnection GetStats() which is no longer blocked on
> encoder initialization.
>
> BUG=webrtc:5410
> R=stefan@webrtc.org
>
> Committed: fb647a67be

R=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410

Review URL: https://codereview.webrtc.org/1821983002 .

Cr-Commit-Position: refs/heads/master@{#12086}
2016-03-22 11:19:14 +00:00
Peter Boström
1d1944187f Replace RefCountImpl with rtc::RefCountedObject.
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.

Also making wider use of scoped_refptr and fixing various leaks in the
process.

BUG=webrtc:5229
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1477013005 .

Cr-Commit-Position: refs/heads/master@{#12075}
2016-03-21 15:44:41 +00:00
Peter Boström
fb647a67be Initialize/configure video encoders asychronously.
Greatly speeds up setRemoteDescription() by moving encoder initialization
off the main worker thread, which is free to move onto gathering ICE
candidates and other tasks while InitEncode() is performed. It also
un-blocks PeerConnection GetStats() which is no longer blocked on
encoder initialization.

BUG=webrtc:5410
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1757313002 .

Cr-Commit-Position: refs/heads/master@{#11983}
2016-03-14 15:59:03 +00:00
kwiberg
27f982bbcb Replace scoped_ptr with unique_ptr in webrtc/video/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1751903002

Cr-Commit-Position: refs/heads/master@{#11833}
2016-03-01 19:52:39 +00:00
danilchap
ac287ee8b5 VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.
render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at.
mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization.

calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged.

BUG=webrtc:5433, webrtc:5504, webrtc:5505

Review URL: https://codereview.webrtc.org/1693443002

Cr-Commit-Position: refs/heads/master@{#11820}
2016-02-29 20:17:10 +00:00
Niels Möller
739fcb989d Cleanup of webrtc::VideoFrame.
Delete EqualsFrame method, used only by tests. Delete one of the
CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
and CopyFrame.

BUG=webrtc:5426

Committed: https://crrev.com/208019637bfed975f8f13b16d40b90e200763cd6
Cr-Commit-Position: refs/heads/master@{#11783}

R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1679323002 .

Cr-Commit-Position: refs/heads/master@{#11811}
2016-02-29 12:11:57 +00:00
kjellander
54ebfca934 Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
Reason for revert:
Breaks downstream compilation. Please make non-breaking API changes for the reland or coordinate fixing downstream code quickly with the sheriff.

Original issue's description:
> Cleanup of webrtc::VideoFrame.
>
> Delete EqualsFrame method, used only by tests. Delete one of the
> CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
> and CopyFrame.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/208019637bfed975f8f13b16d40b90e200763cd6
> Cr-Commit-Position: refs/heads/master@{#11783}

TBR=pbos@webrtc.org,perkj@webrtc.org,pthatcher@webrtc.org,mflodman@webrtc.org,marpan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1743613002

Cr-Commit-Position: refs/heads/master@{#11789}
2016-02-26 15:38:57 +00:00
nisse
208019637b Cleanup of webrtc::VideoFrame.
Delete EqualsFrame method, used only by tests. Delete one of the
CreateFrame methods. Drop return value for CreateEmptyFrame, CreateFrame
and CopyFrame.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1679323002

Cr-Commit-Position: refs/heads/master@{#11783}
2016-02-26 14:40:47 +00:00
Peter Boström
395939781b Reland of Don't send FEC for H.264 with NACK enabled. (patchset #1 id:1 of https://codereview.webrtc.org/1692783005/ )
Reason for revert:
Disabling tests on memcheck that time out due to using real VP8 encoders.

Original issue's description:
> Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ )
>
> Reason for revert:
> Broke the VerifyHistogramStatsWithRed test on the Windows DrMemory Full bot and Linux Memcheck bot. Please fix the test and reland.
>
> Original issue's description:
> > Don't send FEC for H.264 with NACK enabled.
> >
> > The H.264 does not contain picture IDs and are not sufficient to
> > determine that a packet may be skipped. This causes retransmission
> > requests for FEC that are currently dropped by the sender (since they
> > should be redundant).
> >
> > The receiver is then unable to continue without having the packet gap
> > filled (unlike VP8/VP9 which moves on since it has a consecutive stream
> > of picture IDs).
> >
> > Even if FEC retransmission did work it's a huge waste of bandwidth,
> > since it just adds additional overhead that has to be unconditionally
> > transmitted. This bandwidth is better used to send higher-quality
> > frames.
> >
> > BUG=webrtc:5264
> > R=stefan@webrtc.org
> >
> > Committed: https://crrev.com/25558ad819b4df41ba51537e26a77480ace1e525
> > Cr-Commit-Position: refs/heads/master@{#11601}
>
> TBR=stefan@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:5264
>
> Committed: https://crrev.com/29ffdc1a15e31bd81e806ff135c2100d811714f0
> Cr-Commit-Position: refs/heads/master@{#11607}

TBR=stefan@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5264

Review URL: https://codereview.webrtc.org/1697093002 .

Cr-Commit-Position: refs/heads/master@{#11621}
2016-02-15 10:27:22 +00:00
deadbeef
29ffdc1a15 Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ )
Reason for revert:
Broke the VerifyHistogramStatsWithRed test on the Windows DrMemory Full bot and Linux Memcheck bot. Please fix the test and reland.

Original issue's description:
> Don't send FEC for H.264 with NACK enabled.
>
> The H.264 does not contain picture IDs and are not sufficient to
> determine that a packet may be skipped. This causes retransmission
> requests for FEC that are currently dropped by the sender (since they
> should be redundant).
>
> The receiver is then unable to continue without having the packet gap
> filled (unlike VP8/VP9 which moves on since it has a consecutive stream
> of picture IDs).
>
> Even if FEC retransmission did work it's a huge waste of bandwidth,
> since it just adds additional overhead that has to be unconditionally
> transmitted. This bandwidth is better used to send higher-quality
> frames.
>
> BUG=webrtc:5264
> R=stefan@webrtc.org
>
> Committed: https://crrev.com/25558ad819b4df41ba51537e26a77480ace1e525
> Cr-Commit-Position: refs/heads/master@{#11601}

TBR=stefan@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:5264

Review URL: https://codereview.webrtc.org/1692783005

Cr-Commit-Position: refs/heads/master@{#11607}
2016-02-12 20:00:47 +00:00
Peter Boström
25558ad819 Don't send FEC for H.264 with NACK enabled.
The H.264 does not contain picture IDs and are not sufficient to
determine that a packet may be skipped. This causes retransmission
requests for FEC that are currently dropped by the sender (since they
should be redundant).

The receiver is then unable to continue without having the packet gap
filled (unlike VP8/VP9 which moves on since it has a consecutive stream
of picture IDs).

Even if FEC retransmission did work it's a huge waste of bandwidth,
since it just adds additional overhead that has to be unconditionally
transmitted. This bandwidth is better used to send higher-quality
frames.

BUG=webrtc:5264
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1687303002 .

Cr-Commit-Position: refs/heads/master@{#11601}
2016-02-12 14:08:39 +00:00
pbos
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +00:00
Tommi
97888bd95a Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.
While doing this, I made a couple of minor changes:
* Removed unused variables (one lock and one video frame variable)
* Switched over to a scoped lock in remb.cc and removed an if() in a function where we can just return the expression being checked.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1613053003 .

Cr-Commit-Position: refs/heads/master@{#11349}
2016-01-21 22:25:12 +00:00
terelius
429c345b02 Fixes a bug which incorrectly logs incoming RTCP as outgoing.
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1571283002

Cr-Commit-Position: refs/heads/master@{#11336}
2016-01-21 13:42:10 +00:00
asapersson
d9f641e341 Reallocate encoded buffer size if needed. Initially set to the input image size.
Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size.

BUG=chromium:578193

Review URL: https://codereview.webrtc.org/1603643006

Cr-Commit-Position: refs/heads/master@{#11329}
2016-01-21 09:11:43 +00:00
marpan
8432e1f4b8 Re-enable tests that failed under Linux_Msan.
Fixed in latest libvpx roll.
Keep EndToEndTest.TransportSeqNumOnAudioAndVideo disabled on
Win_DrMemory for now as it seems to time-out/too slow.

TBR=stefan@webrtc.org, kjellander@webrtc.org
BUG=webrtc:5402
NOTRY=True

Review URL: https://codereview.webrtc.org/1577313003

Cr-Commit-Position: refs/heads/master@{#11240}
2016-01-13 16:35:51 +00:00
stefan
e74eef19bd Add CreateSend/ReceiveTransport() methods to CallTest.
This allows the test to create its own transports if it, for instance, needs to do demuxing.

BUG=webrtc:5416

Review URL: https://codereview.webrtc.org/1573453002

Cr-Commit-Position: refs/heads/master@{#11187}
2016-01-08 14:47:21 +00:00
Stefan Holmer
9fea80f50d Add audio streams to CallTest and a first A/V call test.
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1542653002 .

Cr-Commit-Position: refs/heads/master@{#11171}
2016-01-07 16:43:31 +00:00
kjellander
44cc795016 Roll chromium_revision 4df108a..2a70cb1 (367307:367468)
Mac 32-bit support has been gone in Chromium for a long time, but was
removed in https://codereview.chromium.org/1557823002. This called
for finally removing our Mac 32-bit builds, which was done in
http://crbug.com/574320.

Change log: 4df108a..2a70cb1
Full diff: 4df108a..2a70cb1

Changed dependencies:
* src/third_party/libvpx_new/source/libvpx: ecb8dff..a9dd8a7
* src/third_party/nss: aee1b12..225bfc3
DEPS diff: 4df108a..2a70cb1/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,
BUG=webrtc:5401, webrtc:5402
NOTRY=True

Review URL: https://codereview.webrtc.org/1556273002

Cr-Commit-Position: refs/heads/master@{#11159}
2016-01-07 06:12:36 +00:00
Peter Boström
13f61dfea5 Move fake-handle frame creation into test target.
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.

Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.

BUG=webrtc:5398
R=kjellander@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1554223002 .

Cr-Commit-Position: refs/heads/master@{#11149}
2016-01-04 21:36:49 +00:00
honghaiz
cfb7f01fd6 Disable VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly
due to flakiness on LinuxAsan.

BUG=webrtc:5382
TBR=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1541923003

Cr-Commit-Position: refs/heads/master@{#11109}
2015-12-21 21:35:00 +00:00
stefan
ff483617a4 Step 1 to prepare call_test.* for combined audio/video tests.
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.

No functional changes.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1537273003

Cr-Commit-Position: refs/heads/master@{#11101}
2015-12-21 11:14:05 +00:00
asapersson
38bb8ad1ca Add test for verifying configured key frame interval for VP9.
BUG=

Review URL: https://codereview.webrtc.org/1498053002

Cr-Commit-Position: refs/heads/master@{#11004}
2015-12-14 09:41:27 +00:00
Peter Boström
5811a39f14 Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.

Does not modify test/channel_transport/.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1487893004 .

Cr-Commit-Position: refs/heads/master@{#10968}
2015-12-10 12:03:00 +00:00
mflodman
d1590b2571 Lint clean video/ and add lint presubmit check.
BUG=webrtc:5316

Review URL: https://codereview.webrtc.org/1507643004

Cr-Commit-Position: refs/heads/master@{#10953}
2015-12-09 15:08:05 +00:00
Stefan Holmer
4654d204e4 Add test which verifies that the RTP header extensions are set correctly for FEC packets.
Also taking the opportunity to do a little bit of clean up.

BUG=webrtc:705
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1506863002 .

Cr-Commit-Position: refs/heads/master@{#10927}
2015-12-08 08:10:58 +00:00
Åsa Persson
ff24c04c73 Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
Specify kf_min_dist to get correct key frame interval in svc mode.

Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).

BUG=chromium:500602
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1492633005 .

Cr-Commit-Position: refs/heads/master@{#10890}
2015-12-04 09:58:23 +00:00