Commit Graph

974 Commits

Author SHA1 Message Date
69dc7dbe24 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
Reason for revert:
This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio

Original issue's description:
> Moving webrtc.gni up one level from build/
>
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2651543003
> Cr-Commit-Position: refs/heads/master@{#16241}
> Committed: 35a32700fc

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2657563002
Cr-Commit-Position: refs/heads/master@{#16244}
2017-01-24 13:14:35 +00:00
35a32700fc Moving webrtc.gni up one level from build/
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2651543003
Cr-Commit-Position: refs/heads/master@{#16241}
2017-01-24 12:49:35 +00:00
62d02c328d Unit test out of band H264 SPS,PPS within RtpStreamReceiver.
This CL introduces a dedicated unit test for webrtc::RtpStreamReceiver.
Focus of this CL is testing RtpStreamReciver::OnReceivedPayloadData().
Dependencies with virtual interfaces are (g)mocked, non-virtual
dependencies are instantiated.

This CL is chained to https://codereview.webrtc.org/2638933002/ .

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2641463002
Cr-Commit-Position: refs/heads/master@{#16240}
2017-01-24 12:38:27 +00:00
d2b092f38a Reland of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
Fixed memory leak in test case.

Original issue's description
> Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ )
>
> Reason for revert:
> Triggers leak on Linux memcheck (non-default trybot):
>
> Original issue's description:
> > H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
> >
> > - Changed method name to clarify that entire Nalus are expected.
> > - Added unit test code.
> > - Adjusted InsetSpsPpsNalus() implementation to above requirement.
> >
> > BUG=webrtc:5948
> >
> > Review-Url: https://codereview.webrtc.org/2638933002
> > Cr-Commit-Position: refs/heads/master@{#16221}
> > Committed: f53d7374cf
>
> TBR=philipel@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5948
> Review-Url: https://codereview.webrtc.org/2649113003
> Cr-Commit-Position: refs/heads/master@{#16225}
> Committed: 914d49d0fd
>

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2651593004
Cr-Commit-Position: refs/heads/master@{#16235}
2017-01-24 10:38:17 +00:00
914d49d0fd Revert of H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header. (patchset #3 id:40001 of https://codereview.webrtc.org/2638933002/ )
Reason for revert:
Triggers leak on Linux memcheck (non-default trybot):

### BEGIN MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)
Command: ../Release/./modules_unittests --isolated-script-test-output=/b/s/w/ioUlJCnu/output.json --isolated-script-test-chartjson-output=/b/s/w/ioUlJCnu/chartjson-output.json --gtest_filter=-CommonFormats/AudioProcessingTest*
Leak_DefinitelyLost
45 bytes in 1 blocks are definitely lost in loss record 118 of 277
  operator new[](unsigned long) (m_replacemalloc/vg_replace_malloc.c:363)
  webrtc::video_coding::H264SpsPpsTracker::CopyAndFixBitstream(webrtc::VCMPacket*) (/b/s/w/irJgAGsR/out/Release/modules_unittests)
  webrtc::video_coding::TestH264SpsPpsTracker_SpsPpsOutOfBand_Test::TestBody() (/b/s/w/irJgAGsR/out/Release/modules_unittests)
Suppression (error hash=#0112A395AF2326BC#):
  For more info on using suppressions see http://dev.chromium.org/developers/tree-sheriffs/sheriff-details-chromium/memory-sheriff#TOC-Suppressing-memory-reports
{
   <insert_a_suppression_name_here>
   Memcheck:Leak
   fun:_Zna*
   fun:_ZN6webrtc12video_coding17H264SpsPpsTracker19CopyAndFixBitstreamEPNS_9VCMPacketE
   fun:_ZN6webrtc12video_coding42TestH264SpsPpsTracker_SpsPpsOutOfBand_Test8TestBodyEv
}
### END MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)

Original issue's description:
> H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
>
> - Changed method name to clarify that entire Nalus are expected.
> - Added unit test code.
> - Adjusted InsetSpsPpsNalus() implementation to above requirement.
>
> BUG=webrtc:5948
>
> Review-Url: https://codereview.webrtc.org/2638933002
> Cr-Commit-Position: refs/heads/master@{#16221}
> Committed: f53d7374cf

TBR=philipel@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2649113003
Cr-Commit-Position: refs/heads/master@{#16225}
2017-01-24 04:16:58 +00:00
f53d7374cf H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
- Changed method name to clarify that entire Nalus are expected.
- Added unit test code.
- Adjusted InsetSpsPpsNalus() implementation to above requirement.

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2638933002
Cr-Commit-Position: refs/heads/master@{#16221}
2017-01-23 17:29:33 +00:00
3626865be2 GN: Refactor modules_unittests to eliminate package boundary violations.
BUG=webrtc:6954

Review-Url: https://codereview.webrtc.org/2629923002
Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 16:27:11 +00:00
04926b8264 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
Reason for revert:
Breaks tests downstream.

Original issue's description:
> Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
>
> Reason for revert:
> Fix in this CL: https://codereview.chromium.org/2640793003/
>
> Original issue's description:
> > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> >
> > Reason for revert:
> > Breaks android bots.
> >
> > Original issue's description:
> > > Make the new jitter buffer the default jitter buffer.
> > >
> > > This CL contains only the changes necessary to make the switch to the new jitter
> > > buffer, clean up will be done in follow up CLs.
> > >
> > > In this CL:
> > >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > >    new video jitter buffer the default one.
> > >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > >
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2627463004
> > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > Committed: 0f0763d86d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2632123005
> > Cr-Commit-Position: refs/heads/master@{#16117}
> > Committed: c08c191f7d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2642753002
> Cr-Commit-Position: refs/heads/master@{#16149}
> Committed: f20dd0014d

TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2638423003
Cr-Commit-Position: refs/heads/master@{#16159}
2017-01-19 08:06:17 +00:00
f20dd0014d Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
Reason for revert:
Fix in this CL: https://codereview.chromium.org/2640793003/

Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
>
> Reason for revert:
> Breaks android bots.
>
> Original issue's description:
> > Make the new jitter buffer the default jitter buffer.
> >
> > This CL contains only the changes necessary to make the switch to the new jitter
> > buffer, clean up will be done in follow up CLs.
> >
> > In this CL:
> >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> >    new video jitter buffer the default one.
> >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> >
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2627463004
> > Cr-Commit-Position: refs/heads/master@{#16114}
> > Committed: 0f0763d86d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2632123005
> Cr-Commit-Position: refs/heads/master@{#16117}
> Committed: c08c191f7d

TBR=stefan@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2642753002
Cr-Commit-Position: refs/heads/master@{#16149}
2017-01-18 15:15:37 +00:00
fcc600651d Clear the FrameBuffer in case of a backward jump in the picture id.
Even though this is against the spec we allow a stream to continue if
a backwards jump in the picture id occurs on a keyframe.

BUG=webrtc:7001, webrtc:5514

Review-Url: https://codereview.webrtc.org/2640793003
Cr-Commit-Position: refs/heads/master@{#16146}
2017-01-18 13:35:20 +00:00
44303ea0ff Revert of Add experimental simulcast screen content mode (patchset #5 id:80001 of https://codereview.webrtc.org/2636443002/ )
Reason for revert:
Breaks chromium.

Original issue's description:
> Add experimental simulcast screen content mode
>
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2636443002
> Cr-Commit-Position: refs/heads/master@{#16135}
> Committed: a28e971e3b

TBR=perkj@webrtc.org,asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2643763002
Cr-Commit-Position: refs/heads/master@{#16145}
2017-01-18 13:19:13 +00:00
a28e971e3b Add experimental simulcast screen content mode
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2636443002
Cr-Commit-Position: refs/heads/master@{#16135}
2017-01-18 08:36:31 +00:00
c08c191f7d Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
Reason for revert:
Breaks android bots.

Original issue's description:
> Make the new jitter buffer the default jitter buffer.
>
> This CL contains only the changes necessary to make the switch to the new jitter
> buffer, clean up will be done in follow up CLs.
>
> In this CL:
>  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
>    new video jitter buffer the default one.
>  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
>    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2627463004
> Cr-Commit-Position: refs/heads/master@{#16114}
> Committed: 0f0763d86d

TBR=stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2632123005
Cr-Commit-Position: refs/heads/master@{#16117}
2017-01-17 12:03:53 +00:00
0f0763d86d Make the new jitter buffer the default jitter buffer.
This CL contains only the changes necessary to make the switch to the new jitter
buffer, clean up will be done in follow up CLs.

In this CL:
 - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
   new video jitter buffer the default one.
 - Moved WebRTC.Video.KeyFramesReceivedInPermille and
   WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2627463004
Cr-Commit-Position: refs/heads/master@{#16114}
2017-01-17 11:31:15 +00:00
57c2fff361 Periodically update channel parameters and send TargetBitrate message.
Currently, parameters are periodically updated, but the TargetBitrate
message is only sent if a new bitrate is set. It should be sent
periodically to indicate the signaled bitrate is valid and to prevent
stale values due to loss of RTCP packets.

BUG=webrtc:6897

Review-Url: https://codereview.webrtc.org/2616393003
Cr-Commit-Position: refs/heads/master@{#16096}
2017-01-16 14:24:02 +00:00
bc5d921659 Rename base/analytics/ to base/numerics/
BUG=webrtc:6832

Review-Url: https://codereview.webrtc.org/2626203002
Cr-Commit-Position: refs/heads/master@{#16060}
2017-01-13 17:14:33 +00:00
71b1c1fc1d Rename fec_tables_xor.h -> fec_rate_table.h
This is a slightly more descriptive name, since we only have one type
of erasure code (XOR), and we only have one table.

BUG=None

Review-Url: https://codereview.webrtc.org/2625903004
Cr-Commit-Position: refs/heads/master@{#16032}
2017-01-12 14:16:24 +00:00
59e99b76da Set |_encodedWidth| and |_encodedHeight| in RtpFrameObject.
BUG=chromium:678080

Review-Url: https://codereview.webrtc.org/2622053003
Cr-Commit-Position: refs/heads/master@{#16029}
2017-01-12 11:26:04 +00:00
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
0ad0de6ef0 Rename incoming_framerate_ to capture_framerate_ in screenshare_layers.
Avoids confusion about the meaning of "incoming".

BUG=webrtc:6897

Review-Url: https://codereview.webrtc.org/2624073003
Cr-Commit-Position: refs/heads/master@{#16007}
2017-01-11 13:01:32 +00:00
ea142f8de3 Don't detect a new frame if a previous packet is used in a previous frame.
In this CL:
 - Removed unused variable |last_seq_num_|.
 - Fixed bug where a new incomplete frame was detected as a complete frame.
 - Added fuzzer to video_coding::PacketBuffer.

BUG=chromium:677101

Review-Url: https://codereview.webrtc.org/2613833003
Cr-Commit-Position: refs/heads/master@{#16003}
2017-01-11 10:01:56 +00:00
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
0d1b2b6880 Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
Add RTC_DEPRACATed anonymous unions to not break downstream projects.

Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date:   Mon Dec 19 09:36:33 2016 -0800
>
>     Revert of Rename RTPVideoHeader.isFirstPacket to
>     .is_first_packet_in_frame. (patchset #1 id:1 of
>     https://codereview.webrtc.org/2574943003/ )
>
>     Reason for revert:
>     breaks downstream project.
>
>     Can you make this change in a compatible way using anonymous
>     union:
>     union {
>       bool is_first_packet_in_frame;
>       RTC_DEPRECATED bool isFirstPacket;
>     };
>     (unfortunetly this this treak breaks braced initialization in
>     rtp_rtcp_impl_unittest.cc,
>     so that should be rewritting in a more classic way)
>
>     Original issue's description:
>     > Rename RTPVideoHeader.isFirstPacket to
>     > .is_first_packet_in_frame.
>     >
>     > Name should represent the actual meaning.
>     >
>     > BUG=None
>     >
>     > Review-Url: https://codereview.webrtc.org/2574943003
>     > Cr-Commit-Position: refs/heads/master@{#15684}
>     > Committed:
>     > efde908380
>
>     TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
>     # Skipping CQ checks because original CL landed less than 1 days
>     ago.
>     NOPRESUBMIT=true
>     NOTREECHECKS=true
>     NOTRY=true
>     BUG=None
>
>     Review-Url: https://codereview.webrtc.org/2589783003
>     Cr-Commit-Position: refs/heads/master@{#15686}
>

BUG=None

Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
2017-01-10 12:21:35 +00:00
2c2f34c1ca PacketBuffer now correctly cast sequence numbers to uint16_t.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2603223002
Cr-Commit-Position: refs/heads/master@{#15885}
2017-01-03 13:55:34 +00:00
c7c26a0e64 Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
Reason for revert:
Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used.

Original issue's description:
> Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
>
> Reason for revert:
> Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
>
> Original issue's description:
> > Replace basictypes.h with stdint.h for int_t types.
> >
> > Removes basictypes.h for types that only makes use of it for fixed-size-int
> > typedefs and replaces it with stdint.h.
> >
> > BUG=webrtc:6853
> > R=tommi@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2604043002
> > Cr-Commit-Position: refs/heads/master@{#15867}
> > Committed: 7fd1a75300
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6853
>
> Review-Url: https://codereview.webrtc.org/2603203003
> Cr-Commit-Position: refs/heads/master@{#15869}
> Committed: 7eb0e23bcf

BUG=webrtc:6853
TBR=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2609783002
Cr-Commit-Position: refs/heads/master@{#15873}
2017-01-02 16:42:32 +00:00
7eb0e23bcf Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
Reason for revert:
Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.

Original issue's description:
> Replace basictypes.h with stdint.h for int_t types.
>
> Removes basictypes.h for types that only makes use of it for fixed-size-int
> typedefs and replaces it with stdint.h.
>
> BUG=webrtc:6853
> R=tommi@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2604043002
> Cr-Commit-Position: refs/heads/master@{#15867}
> Committed: 7fd1a75300

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6853

Review-Url: https://codereview.webrtc.org/2603203003
Cr-Commit-Position: refs/heads/master@{#15869}
2017-01-02 15:32:25 +00:00
7fd1a75300 Replace basictypes.h with stdint.h for int_t types.
Removes basictypes.h for types that only makes use of it for fixed-size-int
typedefs and replaces it with stdint.h.

BUG=webrtc:6853
R=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2604043002
Cr-Commit-Position: refs/heads/master@{#15867}
2017-01-02 14:58:46 +00:00
94b880178a Revert of Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9 (patchset #1 id:1 of https://codereview.webrtc.org/2568743007/ )
Reason for revert:
Trying to re-enable this test as we're now using a newer Clang version (289944-2).

Original issue's description:
> Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
>
> This test is flaky on all platforms, not just Android. Disabling it entirely until webrtc:6057 is fixed.
>
> BUG=webrtc:6057
>
> Committed: https://crrev.com/bb66ec35739830847bfb0146cd029ca41421b2d8
> Cr-Commit-Position: refs/heads/master@{#15594}

TBR=marpan@webrtc.org,sprang@webrtc.org,skvlad@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6057

Review-Url: https://codereview.webrtc.org/2603993002
Cr-Commit-Position: refs/heads/master@{#15821}
2016-12-28 15:22:54 +00:00
ac4a90dfdc Add frame rate throttling to vp8 screenshare_layers.
Drop frames if incoming frame rate is higher than the configured max
framerate.

BUG=webrtc:6897

Review-Url: https://codereview.webrtc.org/2578993002
Cr-Commit-Position: refs/heads/master@{#15819}
2016-12-28 13:58:07 +00:00
8d5608880f Do not call OnDecoderTiming before timing values are set.
Wait until first frame is decoded to avoid include zeros in stats.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2582313002
Cr-Commit-Position: refs/heads/master@{#15752}
2016-12-22 09:26:18 +00:00
022b54e86a Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker.
To avoid making this CL large unittests will be added in a followup CL.

BUG=webrtc:5948

patch from issue 2570073002 at patchset 20001 (http://crrev.com/2570073002#ps20001)

Review-Url: https://codereview.webrtc.org/2565173009
Cr-Commit-Position: refs/heads/master@{#15710}
2016-12-20 12:15:59 +00:00
0ad21111fc Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
Reason for revert:
breaks downstream project.

Can you make this change in a compatible way using anonymous union:
union {
  bool is_first_packet_in_frame;
  RTC_DEPRECATED bool isFirstPacket;
};
(unfortunetly this this treak breaks braced initialization in rtp_rtcp_impl_unittest.cc,
so that should be rewritting in a more classic way)

Original issue's description:
> Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
>
> Name should represent the actual meaning.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2574943003
> Cr-Commit-Position: refs/heads/master@{#15684}
> Committed: efde908380

TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2589783003
Cr-Commit-Position: refs/heads/master@{#15686}
2016-12-19 17:36:33 +00:00
efde908380 Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
Name should represent the actual meaning.

BUG=None

Review-Url: https://codereview.webrtc.org/2574943003
Cr-Commit-Position: refs/heads/master@{#15684}
2016-12-19 16:32:24 +00:00
0cd27ba088 Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
Reason for revert:
Bug affecting perf tests has been fixed. The issue was that I had accidentally disabled cpu overuse adaptation based on the encoders ScalingSettings, not just quality-based scaling.

Original issue's description:
> Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
>
> Reason for revert:
> Breaks perf tests
>
> Original issue's description:
> > Properly report number of quality downscales in stats.
> >
> > A regression was introduced in 876222f that caused these stats to
> > be reported incorrectly. This used to be only implemented for VP8
> > but should now be available for all codecs.
> >
> > BUG=webrtc:6860
> >
> > Review-Url: https://codereview.webrtc.org/2564373002
> > Cr-Commit-Position: refs/heads/master@{#15673}
> > Committed: 0c8c538835
>
> TBR=asapersson@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:6860
>
> Review-Url: https://codereview.webrtc.org/2586783003
> Cr-Commit-Position: refs/heads/master@{#15678}
> Committed: fe04bd43cc

TBR=asapersson@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:6860

Review-Url: https://codereview.webrtc.org/2588743002
Cr-Commit-Position: refs/heads/master@{#15680}
2016-12-19 14:32:16 +00:00
fe04bd43cc Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
Reason for revert:
Breaks perf tests

Original issue's description:
> Properly report number of quality downscales in stats.
>
> A regression was introduced in 876222f that caused these stats to
> be reported incorrectly. This used to be only implemented for VP8
> but should now be available for all codecs.
>
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2564373002
> Cr-Commit-Position: refs/heads/master@{#15673}
> Committed: 0c8c538835

TBR=asapersson@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:6860

Review-Url: https://codereview.webrtc.org/2586783003
Cr-Commit-Position: refs/heads/master@{#15678}
2016-12-19 14:17:30 +00:00
35b41a29b3 Reland of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2585183002/ )
Reason for revert:
Still hitting NOTREACHED.

Original issue's description:
> Revert of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2477663002/ )
>
> Reason for revert:
> To see if the NOTREACHED is still hit.
>
> Original issue's description:
> > Disabling NOTREACHED which we're hitting flakily in browser tests.
> >
> > I have no idea how bad it is that we're hitting this limit; I'm just
> > doing this to stop the tests from flaking.
> >
> > BUG=webrtc:6484
> >
> > Committed: https://crrev.com/6eaa55867b449df992752c1df540ec42f9d9b057
> > Cr-Commit-Position: refs/heads/master@{#14974}
>
> TBR=stefan@webrtc.org,phoglund@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6484
>
> Review-Url: https://codereview.webrtc.org/2585183002
> Cr-Commit-Position: refs/heads/master@{#15665}
> Committed: 9d7ea0920c

TBR=stefan@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6484

Review-Url: https://codereview.webrtc.org/2585273002
Cr-Commit-Position: refs/heads/master@{#15676}
2016-12-19 13:54:15 +00:00
0c8c538835 Properly report number of quality downscales in stats.
A regression was introduced in 876222f that caused these stats to
be reported incorrectly. This used to be only implemented for VP8
but should now be available for all codecs.

BUG=webrtc:6860

Review-Url: https://codereview.webrtc.org/2564373002
Cr-Commit-Position: refs/heads/master@{#15673}
2016-12-19 13:04:40 +00:00
9d7ea0920c Revert of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2477663002/ )
Reason for revert:
To see if the NOTREACHED is still hit.

Original issue's description:
> Disabling NOTREACHED which we're hitting flakily in browser tests.
>
> I have no idea how bad it is that we're hitting this limit; I'm just
> doing this to stop the tests from flaking.
>
> BUG=webrtc:6484
>
> Committed: https://crrev.com/6eaa55867b449df992752c1df540ec42f9d9b057
> Cr-Commit-Position: refs/heads/master@{#14974}

TBR=stefan@webrtc.org,phoglund@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6484

Review-Url: https://codereview.webrtc.org/2585183002
Cr-Commit-Position: refs/heads/master@{#15665}
2016-12-19 07:47:04 +00:00
721d402d71 Create VideoReceiver with external VCMTiming object.
In order for the VCMTiming object to be correctly updated with decoding timings
when running the WebRTC-NewVideoJitterBuffer experiment the VCMTiming object
has to be available in both the VideoReceiver and the video_coding::FrameBuffer
class. Therefore the VCMTiming object is created in VideoRecieveStream and
then passed to VideoReceiver/video_coding::FrameBuffer as they are constructed.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2575473004
Cr-Commit-Position: refs/heads/master@{#15638}
2016-12-15 15:11:01 +00:00
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
e2ec7c270b Remove static cast from H264SpropParameterSets.
This CL is chained to https://codereview.webrtc.org/2539153002/ .

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2568953005
Cr-Commit-Position: refs/heads/master@{#15607}
2016-12-14 14:08:38 +00:00
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
bb66ec3573 Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
This test is flaky on all platforms, not just Android. Disabling it entirely until webrtc:6057 is fixed.

BUG=webrtc:6057

Review-Url: https://codereview.webrtc.org/2568743007
Cr-Commit-Position: refs/heads/master@{#15594}
2016-12-14 09:17:34 +00:00
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
46711db355 Disable flaky QualityScaler tests for now.
BUG=webrtc:6799
TBR=sprang@webrtc.org

Review-Url: https://codereview.webrtc.org/2564423002
Cr-Commit-Position: refs/heads/master@{#15573}
2016-12-13 13:32:31 +00:00
e5ba75a658 Destroy encoders that fail to InitEncode.
SimulcastEncoderAdapter calls Release() on a failed sub-encoder init,
but Release only knows how to clean up encoders that have registered
stream info. Since failed ones don't register, they aren't currently
cleaned up.

BUG=None

Review-Url: https://codereview.webrtc.org/2544003005
Cr-Commit-Position: refs/heads/master@{#15553}
2016-12-12 21:08:36 +00:00
d2ce622ea1 Disabling the potentially flaky test
VideoProcessorIntegrationTest.
ProcessNoLossSpatialResizeFrameDropVP9

TBR=sprang@webrtc.org

BUG=webrtc:6873

Review-Url: https://codereview.webrtc.org/2565373002
Cr-Commit-Position: refs/heads/master@{#15545}
2016-12-12 11:21:21 +00:00
a90799d5fb Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
Reason for revert:
Increase in encode time larger than expected.

Original issue's description:
> Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
>
> Error resilience is currently always enabled for VP9 which reduces quality.
>
> BUG=webrtc:6783
>
> Committed: https://crrev.com/4eb03c76fa2320534d669fda2aabf800e7a6f579
> Cr-Commit-Position: refs/heads/master@{#15390}

TBR=stefan@webrtc.org,marpan@webrtc.org,mflodman@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6783

Review-Url: https://codereview.webrtc.org/2554403006
Cr-Commit-Position: refs/heads/master@{#15501}
2016-12-09 10:35:30 +00:00
36df2d76c5 Refactor webrtc/modules/video_{capture,coding} for GN check
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/video_capture/*",
"//webrtc/modules/video_coding/*",

BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2555333004
Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 17:56:23 +00:00