This CL adds an RTP module to FlexfecReceiveStreamImpl, and wires it up
to send RTCP RRs. It further makes some methods take const refs instead
of values, to make it more clear where packet copies are made. This
change reduces the number of copies by one, for the case when media
packets are added to the FlexFEC receiver.
The end-to-end test is modified to check for RTCP RRs being sent.
Part of this modification involves some indentation changes, and the
diff thus looks bigger than it logically is.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2625633003
Cr-Commit-Position: refs/heads/master@{#16106}
Calling event handlers when the camera is closing is safe because
CameraCapturer checks if the errors are coming from the current session.
Calling onFailure after camera has already started might lead to strange
behavior.
BUG=b/34112992
Review-Url: https://codereview.webrtc.org/2634973002
Cr-Commit-Position: refs/heads/master@{#16101}
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
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}
Based on the mapping between [Audio/Video]TrackInterface and
[Voice/Video][Sender/Receiver]Info.
The IDs of RTCMediaStreamTrackStats are updated to distinguish between
local and remote cases. Previously, if local and remote cases had the
same label only one of them would be included in the report (bug).
BUG=webrtc:6758, chromium:657854, chromium:657855, chromium:657856, chromium:627816
Review-Url: https://codereview.webrtc.org/2610843003
Cr-Commit-Position: refs/heads/master@{#16095}
This information is already available in another member, and
storing it in more places only creates more opportunities
for bugs.
BUG=None
Review-Url: https://codereview.webrtc.org/2613713002
Cr-Commit-Position: refs/heads/master@{#16094}
The MultithreadedFakeH264Encoder is a derived class from FakeEncoder
and FakeH264Encoder, and these should thus also be thread safe.
TESTED=Ran "out/Tsan/video_engine_tests --gtest_filter="*Multithreaded*" --gtest_repeat=100" with is_debug=false, dcheck_always_on=true, is_tsan=true.
BUG=webrtc:6943
Review-Url: https://codereview.webrtc.org/2604403003
Cr-Commit-Position: refs/heads/master@{#16093}
Removed const_cast while creating rtcp packet.
This way manually created packet is as good as parsed packet and can be used in tests directly.
To archive this, changed the way class stores deltas and their sizes:
encoded chunks are stored directly for all but last chunk simplifying rtcp packet creation.
deltas stored together with sequence_number that would allow to simplify reading them from the parsed packet.
Fixed test for maximum received packets.
BUG=None
Review-Url: https://codereview.webrtc.org/2616343003
Cr-Commit-Position: refs/heads/master@{#16091}
This maps, in both directions, [Audio/Video]TrackInterface with
[Voice/Video][Sender/Receiver]Info.
This mapping is necessary for RTCStatsCollector to know the relationship
between RTCMediaStreamTrackStats and RTC[In/Out]boundRTPStreamStats, and
to be able to collect several RTCMediaStreamTrackStats stats.
BUG=webrtc:6757, chromium:659137, chromium:657854, chromium:627816
Review-Url: https://codereview.webrtc.org/2611983002
Cr-Commit-Position: refs/heads/master@{#16090}
The test case is put inside a new test target. That test target will be started from a test script to asses video quality.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2585813002
Cr-Commit-Position: refs/heads/master@{#16088}
Reason for revert:
Broke Chromium FYI bots. See for example https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/8631
FAILED: obj/remoting/protocol/protocol/ice_transport_channel.obj
ninja -t msvc -e environment.x86 -- E:\b\c\cipd\goma/gomacc.exe "E:\b\depot_tools\win_toolchain\vs_files\d3cb0e37bdd120ad0ac4650b674b09e81be45616\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/remoting/protocol/protocol/ice_transport_channel.obj.rsp /c ../../remoting/protocol/ice_transport_channel.cc /Foobj/remoting/protocol/protocol/ice_transport_channel.obj /Fd"obj/remoting/protocol/protocol_cc.pdb"
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2039: 'IceTransportInternal2': is not a member of 'cricket'
e:\b\c\b\win_builder\src\third_party\webrtc\p2p\base\p2ptransportchannel.h(38): note: see declaration of 'cricket'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2065: 'IceTransportInternal2': undeclared identifier
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2065: 'ice_transport': undeclared identifier
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2059: syntax error: 'const'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2143: syntax error: missing ';' before '{'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2447: '{': missing function header (old-style formal list?)
Original issue's description:
> Remove the IceTransportInternal2.
>
> Introduing IceTransportInternal2 is a temporary fix to switch the base
> class of P2PTransportChannel to IceTransportInternal without breaking
> Chromium. It is removed in this CL.
>
> BUG=webrtc:6951
>
> Review-Url: https://codereview.webrtc.org/2632563002
> Cr-Commit-Position: refs/heads/master@{#16083}
> Committed: 8aaa51190eTBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2630243002
Cr-Commit-Position: refs/heads/master@{#16086}
Introduing IceTransportInternal2 is a temporary fix to switch the base
class of P2PTransportChannel to IceTransportInternal without breaking
Chromium. It is removed in this CL.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Commit-Position: refs/heads/master@{#16083}
Also did some slight refactoring of the code that turns constraints
into rtc::Optionals. Used a template method to avoid code duplication,
and used the same pattern for "CopyConstraintsIntoAudioOptions" as was
being used for "CopyConstraintsIntoRtcConfiguration".
BUG=webrtc:6752
Review-Url: https://codereview.webrtc.org/2628523003
Cr-Commit-Position: refs/heads/master@{#16063}
Previously: Failed to setup RTCP mux filter.
Now: rtcpMuxPolicy is 'require', but media description does not
contain 'a=rtcp-mux'.
BUG=webrtc:6966
Review-Url: https://codereview.webrtc.org/2622553003
Cr-Commit-Position: refs/heads/master@{#16062}
That object will be used when we enable RTCP reporting from FlexfecReceiveStream.
Other related changes:
- Stop using FlexfecConfig (from config.h) at receive side in WebRtcVideoEngine2.
- Add a IsCompleteAndEnabled() method to FlexfecReceiveStream::Config, to be
used in WebRtcVideoEngine2.
- Centralize the construction of the FlexfecReceiveStream::Config in unit tests.
This will make future additions to the unit tests cleaner.
- Simplify setup for receiving FlexFEC in VideoQualityTest.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589713003
Cr-Commit-Position: refs/heads/master@{#16059}
Earlier, the FlexFEC codec would receive the same default RTCP feedback
params as the media codecs. Since most of these are not used, there is
no point negotiating them.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2623513002
Cr-Commit-Position: refs/heads/master@{#16057}
A decision has been made to not use prefix header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2590823002
Cr-Commit-Position: refs/heads/master@{#16056}
Just simple "percentage of call that was bad" stats.
BUG=webrtc:6814
Review-Url: https://codereview.webrtc.org/2578213003
Cr-Commit-Position: refs/heads/master@{#16049}
Reading and writing RTP files is implemented elsewhere,
in test/rtp_file_reader.cc and test/rtp_file_writer.cc;
that code is untouched by this cl.
BUG=webrtc:6974
Review-Url: https://codereview.webrtc.org/2633453002
Cr-Commit-Position: refs/heads/master@{#16046}
Previously, BaseChannel supported a "no RTCP" mode, which wasn't
being used any more and is being deleted.
Also, "RTCP mux required" previously worked by calling "ActivateRtcpMux"
after construction. Now it works by explicitly passing a
"require_rtcp_mux" parameter into the constructor.
BUG=None
Review-Url: https://codereview.webrtc.org/2622613004
Cr-Commit-Position: refs/heads/master@{#16045}
The BaseChannel can set the transport directly without depending on
TransportController.
When initializing the network of the BaseChannel, the ChannelManager will
create TransportChannels with the TransportController.
When enabling bundling, WebRtcSession will get or create TransportChannels
with the TransportController.
When a TransportChannel of the BaseChannel needs to be destroyed, it will
fire a signal to notify the WebRtcSession.
BUG=none.
Review-Url: https://codereview.webrtc.org/2614263002
Cr-Commit-Position: refs/heads/master@{#16043}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2608353003
Cr-Commit-Position: refs/heads/master@{#16041}
Ignore CAMERA_ERROR_EVICTED error if camera is about to be closed.
This is valid use case when other app is opening camera while
WebRTC Android app is trying to close it.
BUG=b/34112992
Review-Url: https://codereview.webrtc.org/2627153002
Cr-Commit-Position: refs/heads/master@{#16039}
The video send pipeline uses the magic value 0 for an unset ntp time.
However, the receive pipeline uses the magic value -1 for unset
(unclear where, it seems it behaved differently a few months ago).
This makes cl https://codereview.webrtc.org/2469993003/ fail the
P2PTestConductor.ForwardVideoOnlyStream, because that cl removes code
which always clears the ntp time, and enables propagation of ntp time
from the receive pipeline to the send pipeline.
Treating ntp time <= 0 as unset is a small improvement. Ultimately, a
VideoFrame shouldn't carry an ntp time at all.
BUG=webrtc:5740,webrtc:6977
Review-Url: https://codereview.webrtc.org/2620383005
Cr-Commit-Position: refs/heads/master@{#16035}
We currently leak one local reference to MediaCodecVideoEncoder in
every call to MediaCodecVideoEncoderFactory::CreateVideoEncoder. After
the encoder has been re-initialized 512 times, JNI will crash due to
local reference table overflow (max=512).
The actual leak is in the member initializer list of
MediaCodecVideoEncoder. This CL fixes the leak by adding a
ScopedLocalRefFrame outside of the ctor. All JNI code that originate
from a C++ thread (i.e. the entry point is not a Java thread) must use
a ScopedLocalRefFrame in order to avoid leaking local references.
BUG=webrtc:6969,b/34056152
Review-Url: https://codereview.webrtc.org/2627973004
Cr-Commit-Position: refs/heads/master@{#16034}
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}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
