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f0d1c03c31
The pointer-to-submodule interfaces are being removed. This CL: 1) introduces AudioProcessing::Config::GainController1 with most config, 2) adds functions to APM for setting and getting analog gain, 3) creates a temporary GainControlConfigProxy to support the transition to the new config. 4) Moves the lock references in GainControlForExperimentalAgc and GainControlImpl into the GainControlConfigProxy, as it becomes the sole AGC object with functionality exposed to the client. Bug: webrtc:9947, webrtc:9878 Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27316}
95 lines
2.8 KiB
C++
95 lines
2.8 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/constructor_magic.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(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() const 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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std::unique_ptr<ApmDataDumper> data_dumper_;
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bool enabled_ = false;
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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_;
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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<GainController>> gain_controllers_;
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absl::optional<size_t> num_proc_channels_;
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absl::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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