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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.
100 lines
3.3 KiB
C++
100 lines
3.3 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_GAIN_CONTROL_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_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 "absl/types/optional.h"
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#include "api/array_view.h"
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#include "modules/audio_processing/include/gain_control.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class ApmDataDumper;
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class AudioBuffer;
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class GainControlImpl : public GainControl {
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public:
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GainControlImpl(rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture);
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~GainControlImpl() override;
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void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
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int AnalyzeCaptureAudio(AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
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void Initialize(size_t num_proc_channels, int sample_rate_hz);
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static void PackRenderAudioBuffer(AudioBuffer* audio,
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std::vector<int16_t>* packed_buffer);
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// GainControl implementation.
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bool is_enabled() const override;
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int stream_analog_level() override;
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bool is_limiter_enabled() const override;
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Mode mode() const override;
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int compression_gain_db() const override;
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private:
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class GainController;
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// GainControl implementation.
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int Enable(bool enable) override;
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int set_stream_analog_level(int level) override;
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int set_mode(Mode mode) override;
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int set_target_level_dbfs(int level) override;
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int target_level_dbfs() const override;
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int set_compression_gain_db(int gain) override;
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int enable_limiter(bool enable) override;
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int set_analog_level_limits(int minimum, int maximum) override;
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int analog_level_minimum() const override;
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int analog_level_maximum() const override;
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bool stream_is_saturated() const override;
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int Configure();
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rtc::CriticalSection* const crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_);
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rtc::CriticalSection* const crit_capture_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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bool enabled_ = false;
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Mode mode_ RTC_GUARDED_BY(crit_capture_);
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int minimum_capture_level_ RTC_GUARDED_BY(crit_capture_);
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int maximum_capture_level_ RTC_GUARDED_BY(crit_capture_);
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bool limiter_enabled_ RTC_GUARDED_BY(crit_capture_);
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int target_level_dbfs_ RTC_GUARDED_BY(crit_capture_);
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int compression_gain_db_ RTC_GUARDED_BY(crit_capture_);
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int analog_capture_level_ RTC_GUARDED_BY(crit_capture_);
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bool was_analog_level_set_ RTC_GUARDED_BY(crit_capture_);
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bool stream_is_saturated_ RTC_GUARDED_BY(crit_capture_);
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std::vector<std::unique_ptr<GainController>> gain_controllers_;
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absl::optional<size_t> num_proc_channels_ RTC_GUARDED_BY(crit_capture_);
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absl::optional<int> sample_rate_hz_ RTC_GUARDED_BY(crit_capture_);
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static int instance_counter_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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