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}
The bug hasn't caused us any problems, since we don't run CNG together with Opus (our only real 48 kHz codec), but would cause problems if used with PCB16b @ 48 kHz.
BUG=webrtc:5303
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1496243002 .
Cr-Commit-Position: refs/heads/master@{#10929}
This is needed so that the data channel isn't deleted while one of its
own methods is on the call stack.
BUG=565048
Review URL: https://codereview.webrtc.org/1492383002
Cr-Commit-Position: refs/heads/master@{#10923}
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1457383002 .
Cr-Commit-Position: refs/heads/master@{#10921}
Brings in fixes that save log output to disk instead of piping them
through Python. Should fix problem where output from tests stall for
more than 10 seconds.
Also enabling JsepPeerConnectionP2PTestClient on all platforms again.
BUG=webrtc:5231
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1509463002 .
Cr-Commit-Position: refs/heads/master@{#10917}
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
