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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
bjornv@webrtc.org0c6f9312012-01-30 09:39:08 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
bjornv@webrtc.org0c6f9312012-01-30 09:39:08 +000011#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
12#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
niklase@google.com470e71d2011-07-07 08:21:25 +000013
14#include <vector>
15
pbos@webrtc.org7fad4b82013-05-28 08:11:59 +000016#include "webrtc/modules/audio_processing/include/audio_processing.h"
17#include "webrtc/modules/audio_processing/processing_component.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000018
19namespace webrtc {
andrew@webrtc.org56e4a052014-02-27 22:23:17 +000020
niklase@google.com470e71d2011-07-07 08:21:25 +000021class AudioBuffer;
andrew@webrtc.org56e4a052014-02-27 22:23:17 +000022class CriticalSectionWrapper;
niklase@google.com470e71d2011-07-07 08:21:25 +000023
24class GainControlImpl : public GainControl,
25 public ProcessingComponent {
26 public:
andrew@webrtc.org56e4a052014-02-27 22:23:17 +000027 GainControlImpl(const AudioProcessing* apm,
28 CriticalSectionWrapper* crit);
niklase@google.com470e71d2011-07-07 08:21:25 +000029 virtual ~GainControlImpl();
30
31 int ProcessRenderAudio(AudioBuffer* audio);
32 int AnalyzeCaptureAudio(AudioBuffer* audio);
33 int ProcessCaptureAudio(AudioBuffer* audio);
34
35 // ProcessingComponent implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000036 int Initialize() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000037
38 // GainControl implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000039 bool is_enabled() const override;
40 int stream_analog_level() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000041
42 private:
43 // GainControl implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000044 int Enable(bool enable) override;
45 int set_stream_analog_level(int level) override;
46 int set_mode(Mode mode) override;
47 Mode mode() const override;
48 int set_target_level_dbfs(int level) override;
49 int target_level_dbfs() const override;
50 int set_compression_gain_db(int gain) override;
51 int compression_gain_db() const override;
52 int enable_limiter(bool enable) override;
53 bool is_limiter_enabled() const override;
54 int set_analog_level_limits(int minimum, int maximum) override;
55 int analog_level_minimum() const override;
56 int analog_level_maximum() const override;
57 bool stream_is_saturated() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000058
59 // ProcessingComponent implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000060 void* CreateHandle() const override;
61 int InitializeHandle(void* handle) const override;
62 int ConfigureHandle(void* handle) const override;
63 void DestroyHandle(void* handle) const override;
64 int num_handles_required() const override;
65 int GetHandleError(void* handle) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000066
andrew@webrtc.org56e4a052014-02-27 22:23:17 +000067 const AudioProcessing* apm_;
68 CriticalSectionWrapper* crit_;
niklase@google.com470e71d2011-07-07 08:21:25 +000069 Mode mode_;
70 int minimum_capture_level_;
71 int maximum_capture_level_;
72 bool limiter_enabled_;
73 int target_level_dbfs_;
74 int compression_gain_db_;
75 std::vector<int> capture_levels_;
76 int analog_capture_level_;
77 bool was_analog_level_set_;
78 bool stream_is_saturated_;
79};
80} // namespace webrtc
81
bjornv@webrtc.org0c6f9312012-01-30 09:39:08 +000082#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_