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}
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}
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}
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}
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}
This CL is the first in a series of CLs to refactor
VideoProcessing(Module) to follow Google C++ style guide and make the
code more readable.
This CL removed inheritance from Module, renames variables and makes
VideoProcessingImpl::PreprocessFrame return a frame pointer if there
is a frame to send, nullptr otherwise. The affected CLs also passes git
cl lint.
BUG=webrtc:5259
Review URL: https://codereview.webrtc.org/1482913003
Cr-Commit-Position: refs/heads/master@{#10907}
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
TBR=pthatcher@webrtc.org
BUG=webrtc:3618
This is a reland of https://codereview.webrtc.org/1453523002
Review URL: https://codereview.webrtc.org/1505573002 .
Cr-Commit-Position: refs/heads/master@{#10903}
This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63.
This commit somehow is different from what I have in my local copy. Revert and will recommit.
TBR=pthatcher@webrtc.org
BUG=3618
Review URL: https://codereview.webrtc.org/1494373004 .
Cr-Commit-Position: refs/heads/master@{#10902}
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
BUG=webrtc:3618
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1453523002 .
Cr-Commit-Position: refs/heads/master@{#10901}
and make it configurable from the app.
Changed the decision on whether a connection is pingable:
1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection.
2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate.
Note the default behavior is the same as before.
Also cached the channel state since we are accessing it more often.
BUG=webrtc:5034
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1455033004 .
Cr-Commit-Position: refs/heads/master@{#10900}
Also stop it if the request timed out.
It is going to be complicated to keep this and make it sync with the connection bind request as they may be on two different ports.
BUG=
Review URL: https://codereview.webrtc.org/1465843004
Cr-Commit-Position: refs/heads/master@{#10899}
- Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent().
- Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
BUG=webrtc:4690
R=pthatcher@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1491743004 .
Cr-Commit-Position: refs/heads/master@{#10895}
Specify kf_min_dist to get correct key frame interval in svc mode.
Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).
BUG=chromium:500602
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1492633005 .
Cr-Commit-Position: refs/heads/master@{#10890}
This CL duplicates all the histograms in SendStatisticsProxy. Might be
overkill, but we don't know which stats will be interesting and it makes
the change easier.
BUG=
Review URL: https://codereview.webrtc.org/1433393002
Cr-Commit-Position: refs/heads/master@{#10885}
To try to resolve the problem I replaced the custom synchronization with rtc::Event which made the code cleaner, faster, and less error prone.
However, in the end the source of the test locks was that during TearDown one of the threads was stuck in a waiting loop.
I added a fix for the TearDown issue but still decided to keep the rtc:Event - based code change metioned above as that gave a more clean code.
BUG=
Review URL: https://codereview.webrtc.org/1490113004
Cr-Commit-Position: refs/heads/master@{#10880}
Check if it is in the list of turn entries before attempting to delete it.
BUG=
Review URL: https://codereview.webrtc.org/1458013004
Cr-Commit-Position: refs/heads/master@{#10877}
The two added macros simplifies the logging code when a value which is not stored in a variable should be logged.
BUG=
Review URL: https://codereview.webrtc.org/1488613002
Cr-Commit-Position: refs/heads/master@{#10870}
