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https://github.com/mollyim/webrtc.git
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080006b42f
More robust API option that allows to fully initialize TS when created. Bug: webrtc:13663 Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36490}
36 lines
1.1 KiB
C++
36 lines
1.1 KiB
C++
/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/optionally_built_submodule_creators.h"
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#include <memory>
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#include "modules/audio_processing/transient/transient_suppressor_impl.h"
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namespace webrtc {
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std::unique_ptr<TransientSuppressor> CreateTransientSuppressor(
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const ApmSubmoduleCreationOverrides& overrides,
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TransientSuppressor::VadMode vad_mode,
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int sample_rate_hz,
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int detection_rate_hz,
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int num_channels) {
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#ifdef WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR
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return nullptr;
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#else
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if (overrides.transient_suppression) {
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return nullptr;
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}
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return std::make_unique<TransientSuppressorImpl>(
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vad_mode, sample_rate_hz, detection_rate_hz, num_channels);
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#endif
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}
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} // namespace webrtc
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