b5c48b97f6
Ongoing fixes and improvements, transient suppressor is gone. Also, dropping isac because it doesn't seem to be useful, and is just build system deadweight now. Upstream references: Version: 131.0.6778.200 WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82 Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
92 lines
2.9 KiB
C++
92 lines
2.9 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 <optional>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/agc/gain_control.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();
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GainControlImpl(const GainControlImpl&) = delete;
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GainControlImpl& operator=(const GainControlImpl&) = delete;
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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(const 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(const AudioBuffer& audio,
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std::vector<int16_t>* packed_buffer);
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// GainControl implementation.
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int stream_analog_level() const override;
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bool is_limiter_enabled() const override { return limiter_enabled_; }
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Mode mode() const override { return mode_; }
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int set_mode(Mode mode) override;
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int compression_gain_db() const override { return compression_gain_db_; }
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int set_analog_level_limits(int minimum, int maximum) override;
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int set_compression_gain_db(int gain) override;
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int set_target_level_dbfs(int level) override;
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int enable_limiter(bool enable) override;
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int set_stream_analog_level(int level) override;
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private:
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struct MonoAgcState;
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// GainControl implementation.
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int target_level_dbfs() const override { return target_level_dbfs_; }
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int analog_level_minimum() const override { return minimum_capture_level_; }
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int analog_level_maximum() const override { return maximum_capture_level_; }
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bool stream_is_saturated() const override { return stream_is_saturated_; }
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int Configure();
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std::unique_ptr<ApmDataDumper> data_dumper_;
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Mode mode_;
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int minimum_capture_level_;
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int maximum_capture_level_;
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bool limiter_enabled_;
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int target_level_dbfs_;
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int compression_gain_db_;
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int analog_capture_level_ = 0;
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bool was_analog_level_set_;
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bool stream_is_saturated_;
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std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
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std::vector<int> capture_levels_;
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std::optional<size_t> num_proc_channels_;
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std::optional<int> sample_rate_hz_;
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static int instance_counter_;
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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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