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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
40 lines
1.2 KiB
C++
40 lines
1.2 KiB
C++
/*
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* Copyright (c) 2018 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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#ifndef API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
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#define API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
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#include <memory>
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#include "api/audio/echo_canceller3_config.h"
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#include "api/audio/echo_control.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory {
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public:
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// Factory producing EchoCanceller3 instances with the default configuration.
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EchoCanceller3Factory();
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// Factory producing EchoCanceller3 instances with the specified
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// configuration.
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explicit EchoCanceller3Factory(const EchoCanceller3Config& config);
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// Creates an EchoCanceller3 running at the specified sampling rate.
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std::unique_ptr<EchoControl> Create(int sample_rate_hz) override;
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private:
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const EchoCanceller3Config config_;
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};
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} // namespace webrtc
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#endif // API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
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