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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
99 lines
3.3 KiB
C++
99 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/constructor_magic.h"
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#include "rtc_base/critical_section.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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