Sparse macro is replaced and new implementation in metrics.h is used.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1564923008
Cr-Commit-Position: refs/heads/master@{#11483}
This CL fixes an issue where the "writable" flag didn't stay set after
::send or ::sendto only sent a partial buffer.
Also SocketTest::TcpInternal has been updated to use rtc::Buffer instead
of manually allocating data.
BUG=webrtc:4898
Review URL: https://codereview.webrtc.org/1616153007
Cr-Commit-Position: refs/heads/master@{#11480}
There is a use case with external codec factories that only support
encoding but not decoding for a given type. This leads to a crash
due to null being registered as codec (after a DCHECK).
This CL adds a NullVideoDecoder that is used instead of the null to
not crash but log to LS_ERROR.
BUG=webrtc:5249
Review URL: https://codereview.webrtc.org/1657023002
Cr-Commit-Position: refs/heads/master@{#11475}
Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name.
The idea is to be able to turn off H264 from chromium with this function because...
1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled.
2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable.
3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not.
The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory.
BUG=chromium:500605, chromium:468365
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1657273002
Cr-Commit-Position: refs/heads/master@{#11474}
Adds negotiation of rtx codecs for red and vp9. To keep backwards
compatibility with older Chrome versions, this change includes two
hacks:
1. Red packets will be retransmitted over the rtx codec associated with
vp8 if no rtx codec is associated with red. This is how Chrome does
it today and ensures that we still can send red over rtx to older
versions.
2. If rtx packets associated with the media codec (vp8/vp9 etc) are
received and red has been negotiated, we will assume that the sender
incorrectly has packetized red inside the rtx header associated with
media. We will therefore restore it with the red payload type
instead, which ensures that we can still receive rtx associated with
red from old versions.
Offering multiple rtx codecs to older versions should not be a problem
since old versions themselves only try to negotiate rtx for vp8.
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:4024
TEST=Verified by running apprtc and emulating packet loss between Chrome with and without the patch.
Review URL: https://codereview.webrtc.org/1649493004 .
Cr-Commit-Position: refs/heads/master@{#11472}
It's generated by some encoders between SPS/PPS and an IDR frame, so we should treat it like sps/pps.
BUG=
Review URL: https://codereview.webrtc.org/1664733002
Cr-Commit-Position: refs/heads/master@{#11470}
* SSRCDatabase doesn't need to be a global instance, so I've changed it to be a "regular" class (i.e. construct via ctor, not maybe via GetSSRCDatabase( + release via ReturnSSRCDatabase())). If we ever have parallel tests running in the same process, they won't have the problem of using the same ssrc database.
* Made RtpSender a more const. Also added some todos for myself and holmer to look into clarifying the threading model.
* Switched from CriticalSectionWrapper to rtc::CriticalSection
* Changed the random seeding to use TickTime::Now().Ticks() since TimeInMicroseconds() could return 0 when the process was starting. This is what TimeInMicroseconds() does anyway but now we don't need to access a global clock object.
BUG=webrtc:3062
Review URL: https://codereview.webrtc.org/1623543002
Cr-Commit-Position: refs/heads/master@{#11462}
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.
In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.
BUG=chromium:500605, chromium:468365, webrtc:5427
Review URL: https://codereview.webrtc.org/1639273002
Cr-Commit-Position: refs/heads/master@{#11456}
For example, when the TURN port has an ALLOCATE_MISMATCH error.
BUG=webrtc:5432
Review URL: https://codereview.webrtc.org/1595613004
Cr-Commit-Position: refs/heads/master@{#11453}
Some WebRTC client had a problem with the change "Removing webrtc::AudioFrame::energy_". Now it is solved.
This reverts commit 2bdcfadc8abd418a30dd5cdf54ba45a429d3d9bf.
BUG=webrtc:3315
Review URL: https://codereview.webrtc.org/1638553003
Cr-Commit-Position: refs/heads/master@{#11448}
A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1652983002
Cr-Commit-Position: refs/heads/master@{#11447}
Reason for revert:
May be the reason for mac_asan timeout
Original issue's description:
> Changed test to validate rtp timstamps not just in RTP packets but also in RTCP Sender Reports.
> Altered it to accept negative value since it is normal for RTCP packet coming before RTP packet to have slightly later time.
>
> BUG=webrtc:5433
>
> Committed: https://crrev.com/f4b9c775122b463db7eb2c4101603759a0d00caf
> Cr-Commit-Position: refs/heads/master@{#11417}
TBR=pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5433
Review URL: https://codereview.webrtc.org/1652973002
Cr-Commit-Position: refs/heads/master@{#11446}
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.
Also moves sending transport feedback to the pacer thread.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1628683002
Cr-Commit-Position: refs/heads/master@{#11443}
We have seen an instance of flakiness of the perf tests where it looked
like timestamp wraparound could be an issue. Better safe...
BUG=
Review URL: https://codereview.webrtc.org/1645463002
Cr-Commit-Position: refs/heads/master@{#11440}
Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast.
Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000
Hiding snprintf definition if building with Visual Studio 2015
Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc.
BUG=webrtc:5183
Review URL: https://codereview.webrtc.org/1412653006
Cr-Commit-Position: refs/heads/master@{#11434}
If it still handle packets, when a ping arrives, it will pass the packet to p2ptransportchannel, eventually causing an ASSERT error there (when p2ptransportchannel tries to create a connection from the ping request from unknown address).
BUG=
Review URL: https://codereview.webrtc.org/1649493006
Cr-Commit-Position: refs/heads/master@{#11430}
It's never used anywhere, so it only causes confusion between
itself and SessionDescriptionInterface::candidates.
Review URL: https://codereview.webrtc.org/1642733002
Cr-Commit-Position: refs/heads/master@{#11420}
This argument is never used as a reference and the pointer that's bound
to the const reference may be nullptr. This is undefined behavior and
barks under UBSan.
BUG=webrtc:5124
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1642863003 .
Cr-Commit-Position: refs/heads/master@{#11418}
Altered it to accept negative value since it is normal for RTCP packet coming before RTP packet to have slightly later time.
BUG=webrtc:5433
Review URL: https://codereview.webrtc.org/1633843003
Cr-Commit-Position: refs/heads/master@{#11417}
The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
And the list goes on, there's a dozen of different classes which act as video frame sinks.
At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
BUG=webrtc:5426
R=perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
Cr-Commit-Position: refs/heads/master@{#11396}
Review URL: https://codereview.webrtc.org/1594973006
Cr-Commit-Position: refs/heads/master@{#11414}
The level of the error signal after linear echo cancellation was based on non-buffered signal while that of the near-end and far-end signal based on buffered signal. This discrepancy made the comparison of them unfair.
This CL is to make calculating the error level rely on the same buffering.
BUG=
Review URL: https://codereview.webrtc.org/1510873004
Cr-Commit-Position: refs/heads/master@{#11408}
Before this fix, it was required that the EGL context used as an argument was kept open until all PeerConnections using the context had been closed. With this fix, that is no longer required.
Also, if a released EGLContext (EGL_NO_CONTEXT) is used, harware codecs will fallback to use byte buffers for encoding and decoding.
BUG=b/26583522
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1615153002 .
Cr-Commit-Position: refs/heads/master@{#11398}
Reason for revert:
Broke chrome build. Investigating.
First error relating to AddSink method in mock_peer_connection_dependency_factory.h
Original issue's description:
> New rtc::VideoSinkInterface.
>
> The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
>
> And the list goes on, there's a dozen of different classes which act as video frame sinks.
>
> At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
>
> BUG=webrtc:5426
> R=perkj@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
> Cr-Commit-Position: refs/heads/master@{#11396}
TBR=pthatcher@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:5426
Review URL: https://codereview.webrtc.org/1646463002
Cr-Commit-Position: refs/heads/master@{#11397}
