Reason for revert:
Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
Original issue's description:
> Free SCTP data channels asynchronously in PeerConnection.
>
> This is needed so that the data channel isn't deleted while one of its
> own methods is on the call stack.
>
> BUG=565048
>
> Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> Cr-Commit-Position: refs/heads/master@{#10923}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=565048
Review URL: https://codereview.webrtc.org/1513143003
Cr-Commit-Position: refs/heads/master@{#10977}
Fixes one sign mismatch warning, and one "const has no effect and is
ignored" warning.
BUG=chromium:567877
Review URL: https://codereview.webrtc.org/1510233002
Cr-Commit-Position: refs/heads/master@{#10976}
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.
Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.
Review URL: https://codereview.webrtc.org/1428293003
Cr-Commit-Position: refs/heads/master@{#10974}
By reducing the length of the audio input, the total runtime of
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.*
is reduced by more than 10x, when run single-threaded.
The PCMFile helper class is extended with a FastForward method (to
skip initial silence in the test files) and a limiter on how much to
read.
BUG=webrtc:2463
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1513223002 .
Cr-Commit-Position: refs/heads/master@{#10973}
do the conversion using an opengl fragment shader.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1460703002
Cr-Commit-Position: refs/heads/master@{#10972}
They were meant to be run if we have either iSAC float or fix, but the
typo made them run for just float.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1513483005
Cr-Commit-Position: refs/heads/master@{#10969}
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}
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.
BUG=webrtc:1600
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506103006 .
Cr-Commit-Position: refs/heads/master@{#10964}
-Moved memsets to where their variables are used.
-Removed redundant.
-Changed a pointer scalar to be accessed in pointer notation rather than
in array notation.
The change has been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494473006
Cr-Commit-Position: refs/heads/master@{#10963}
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
-Moved filter reset from the echo suppression
into the echo subtraction code where it belongs
(the echo subtractor should own its filter reset).
-Moved the selection between using the microphone sinal and
the echo subtractor output down to the lowest level in the
EchoSuppression function. This makes sense as that selection
was very hidden in an unrelated sub-sub-function call and
as the selection is critical for what the AEC outputs.
The changes have been tested for bitexactness.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1499573003
Cr-Commit-Position: refs/heads/master@{#10956}
This is needed for Chromium so that we can roll, update libjingle.gyp and then continue.
BUG=chromium:121673
Review URL: https://codereview.webrtc.org/1514573003
Cr-Commit-Position: refs/heads/master@{#10955}
to simplify future refactoring and development.
In more detail:
1) Moved the updating of eBuf from the EchoSubtraction method
to the EchoSuppression method as it is only used in the latter.
2) Moved the computation of efw and dfw from the SubbandCoherence method
as those are actually the analysis filterbank computation that is not
directly related to the coherence.
3) As a consequence of 2) 3 functions needed to be replaced by the
generic function pointer scheme used in WebRTCAec as they have
optimized versions for SSE2 and NEON (which before were local to each
of the aec_core*.c files.
Motivation:
Apart from making sense from a logical point of view, the changes will
a) Allow eBuf stored in half the size on the state.
b) Allow simpler switching between using the the microphone signal
and echo subtractor output in the echo suppressor.
c) Allow further refactoring that move all the changes to eBuf to one method
(currently those are happening in at least 4 different methods.
Drawbacks:
i) dfw is moved to EchoSuppression which increases the stack usage for that
method. This will, however, be improved once further refactoring can be done.
The changes have been tested for bitexactness on Linux using a quite extensive dataset.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1494563002
Cr-Commit-Position: refs/heads/master@{#10954}
The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1484343003
Cr-Commit-Position: refs/heads/master@{#10952}
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.
BUG=webrtc:5318
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512853002 .
Cr-Commit-Position: refs/heads/master@{#10947}
If a description is set that requires making a default stream, and one
already exists, we'll now keep the existing default audio/video tracks,
rather than destroying them and recreating them. Destroying them caused
the blink MediaStream to go to an "ended" state, which is the root cause
of the bug.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1469833006
Cr-Commit-Position: refs/heads/master@{#10946}
Adds tracing specifically to Close, for creating streams and also moves
tracing for SetLocal/RemoteDescription from WebRtcSession. Also adding
some tracing in ChannelManager to see what's taking time inside Close.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1509903002 .
Cr-Commit-Position: refs/heads/master@{#10943}
This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1502223002
Cr-Commit-Position: refs/heads/master@{#10941}
This applies to AcmSwitchingOutputFrequencyOldApi.*,
AcmReceiverBitExactnessOldApi.* and AcmSenderBitExactnessOldApi.*.
BUG=webrtc:4647
NOTRY=true
Review URL: https://codereview.webrtc.org/1503043003
Cr-Commit-Position: refs/heads/master@{#10936}
Also removes virtual from VideoDecoder::Decode and updated mocks and
tests accordingly to use VideoDecoder::DecodeInternal instead.
BUG=webrtc:5167
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1512483003 .
Cr-Commit-Position: refs/heads/master@{#10935}
Adds tracing to WebRtcSession and corresponding BaseChannel calls to see
where time is spent better.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1505023003 .
Cr-Commit-Position: refs/heads/master@{#10934}
This change adds fuzzer tests for iLBC, iSAC fix and float, and
Opus. The fuzzer function takes a random input vector and splits it
into a number of payloads. The lengths of the payloads is also
determined by the random vector. The payloads are decoded with the
decoders.
BUG=webrtc:5306
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1499093002 .
Cr-Commit-Position: refs/heads/master@{#10932}
