/*
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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 <vector>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_audio/swap_queue.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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namespace webrtc {
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class AudioBuffer;
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class GainControlImpl : public GainControl,
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public ProcessingComponent {
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public:
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GainControlImpl(const AudioProcessing* apm,
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rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture);
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virtual ~GainControlImpl();
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int ProcessRenderAudio(AudioBuffer* audio);
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int AnalyzeCaptureAudio(AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio);
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// ProcessingComponent implementation.
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int Initialize() override;
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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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// Reads render side data that has been queued on the render call.
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void ReadQueuedRenderData();
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private:
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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 compression_gain_db() const 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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// ProcessingComponent implementation.
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void* CreateHandle() const override;
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int InitializeHandle(void* handle) const override;
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int ConfigureHandle(void* handle) const override;
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void DestroyHandle(void* handle) const override;
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size_t num_handles_required() const override;
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int GetHandleError(void* handle) const override;
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void AllocateRenderQueue();
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// Not guarded as its public API is thread safe.
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const AudioProcessing* apm_;
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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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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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std::vector<int> capture_levels_ 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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size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
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GUARDED_BY(crit_capture_);
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std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
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std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
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// Lock protection not needed.
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rtc::scoped_ptr<
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SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
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render_signal_queue_;
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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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