Suppression filter is extended to support the synthesis
of multiple channels. This CL is also a major clean-up of ApplyGain.
The CL has been tested for bit-exactness for single channel output.
Bug: webrtc:10913
Change-Id: I1319f127981552e17dec66701a248d34dcf0e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154341
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29284}
Callback set by HardwareVideoEncoder, and wired to the codec's
releaseOutputBuffer. Intention is to move call of this method to the
destructor of a corresponding C++ class in a followup cl, and
eliminate an allocation and memcpy in the process.
Bug: webrtc:9378
Change-Id: I578480b63b68e6ac7a96cdde36379b3c50f05c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142160
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29283}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
"s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"
Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
MSVC only defines _M_ARM64 to identify Windows ARM64, so detect this macro is
necessary to building WebRTC with MSVC.
Bug: chromium:893460
Change-Id: Idd49dfb6dcf367d5efdec705b84db55087de37c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154281
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29280}
Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.
Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}
Mainly an update of the BasicPortAllocator tests. Other code related
to gturn may be deleted in a followup cl.
Bug: None
Change-Id: I72146c8faf30c1a9d03d320d75af91984f797a7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153485
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29275}
Transport parameters are no longer retreived using this method, and no
implementations currently override it.
Bug: webrtc:9719
Change-Id: Iba0e1c7a320266f199aab6f2add36c6a22b48458
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154004
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29272}
https://webrtc-review.googlesource.com/c/src/+/152740 changed the way
OnRtcpPacketReceived is invoked, so that Channel no longer handles the
hop to the worker thread internally. This change updates the test to
hop to the worker thread before calling channel, which fixes a test-only
tsan failure.
Bug: webrtc:10983
Change-Id: Ia31920791fc6eeee86c0d59aa091d708d706bcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154244
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29271}
Echo remover processes all microphone signals. Suppression gains are
computed separately for each capture signal. The minimum gains determine
the final suppression gains applied.
Only the first channel is synthesized. A follow-up CL will add the
synthesis of the remaining channels.
Bug: webrtc:10913
Change-Id: Ife7e74c9a9c6c208fca3992e3cfa840b6b7afcfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153526
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29269}
The high bitrate smoketest is flaky on some platforms,
this CL reduces the resolution and bitrates to make it less
flaky.
Bug: webrtc:10975
Change-Id: Id271b3c68abfa2011c207e7883cfcb230b1d3e36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153845
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29268}
PeerConnection now watches when data channels become ready to send
through its implementation of DataChannelSink, and no longer needs to
monitor the MediaTransport state.
Bug: webrtc:9719
Change-Id: I3e17747eb03926a3791c204bf5a1d2dc67855c09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154001
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29261}
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.
Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
Returning absl::string_view causes problems to the Chromium/WebRTC
component build because absl::operator<< needs to be exported.
This CL switches to `const char*` which should be enough to avoid
to generate temporaries.
Bug: webrtc:9419
Change-Id: If169a6f95c7efd21ac8ce108c7f2f80a76ff2313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153842
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29250}
Currently, APM fakes multichannel in two ways:
- With injected AECs, capture processing is only performed on the left
channel. The result is copied into the other channels.
- With multichannel render audio, all channels are mixed into one
before analysing.
This CL adds a flag to disable these behaviors, ensuring proper
multichannel processing happens throughout the APM pipeline.
Adds killswitches to separately disable render / capture multichannel.
Additionally - AEC3 currently crashes when running with multichannel.
This CL adds the missing pieces to at least have it run without
triggering any DCHECKS, including making the high pass filter properly
handle multichannel.
Bug: webrtc:10913, webrtc:10907
Change-Id: I38795bf8f312b959fcc816a056fba2c68d4e424d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29248}
Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.
Only supported on Android N and higher.
Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.
Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.
See go/webrtc-adm-android for more details and examples.
Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29236}
