c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
98 lines
3.3 KiB
C++
98 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 WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/swap_queue.h"
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#include "webrtc/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_ 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_ GUARDED_BY(crit_capture_);
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int minimum_capture_level_ GUARDED_BY(crit_capture_);
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int maximum_capture_level_ GUARDED_BY(crit_capture_);
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bool limiter_enabled_ GUARDED_BY(crit_capture_);
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int target_level_dbfs_ GUARDED_BY(crit_capture_);
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int compression_gain_db_ GUARDED_BY(crit_capture_);
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int analog_capture_level_ GUARDED_BY(crit_capture_);
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bool was_analog_level_set_ GUARDED_BY(crit_capture_);
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bool stream_is_saturated_ GUARDED_BY(crit_capture_);
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std::vector<std::unique_ptr<GainController>> gain_controllers_;
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rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
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rtc::Optional<int> sample_rate_hz_ 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 // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
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