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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.
89 lines
2.7 KiB
C++
89 lines
2.7 KiB
C++
/*
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* Copyright (c) 2012 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 MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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namespace webrtc {
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class AudioBuffer;
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// The acoustic echo control for mobile (AECM) component is a low complexity
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// robust option intended for use on mobile devices.
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class EchoControlMobileImpl {
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public:
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EchoControlMobileImpl();
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~EchoControlMobileImpl();
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int Enable(bool enable);
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bool is_enabled() const;
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// Recommended settings for particular audio routes. In general, the louder
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// the echo is expected to be, the higher this value should be set. The
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// preferred setting may vary from device to device.
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enum RoutingMode {
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kQuietEarpieceOrHeadset,
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kEarpiece,
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kLoudEarpiece,
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kSpeakerphone,
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kLoudSpeakerphone
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};
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// Sets echo control appropriate for the audio routing |mode| on the device.
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// It can and should be updated during a call if the audio routing changes.
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int set_routing_mode(RoutingMode mode);
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RoutingMode routing_mode() const;
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// Comfort noise replaces suppressed background noise to maintain a
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// consistent signal level.
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int enable_comfort_noise(bool enable);
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bool is_comfort_noise_enabled() const;
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void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
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int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
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void Initialize(int sample_rate_hz,
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size_t num_reverse_channels,
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size_t num_output_channels);
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static void PackRenderAudioBuffer(const AudioBuffer* audio,
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size_t num_output_channels,
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size_t num_channels,
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std::vector<int16_t>* packed_buffer);
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static size_t NumCancellersRequired(size_t num_output_channels,
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size_t num_reverse_channels);
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private:
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class Canceller;
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struct StreamProperties;
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int Configure();
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bool enabled_ = false;
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RoutingMode routing_mode_;
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bool comfort_noise_enabled_;
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std::vector<std::unique_ptr<Canceller>> cancellers_;
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std::unique_ptr<StreamProperties> stream_properties_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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