• Home
  • Line#
  • Scopes#
  • Navigate#
  • Raw
  • Download
1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
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 
11 #ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
12 #define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
13 
14 #include <stddef.h>
15 #include <stdint.h>
16 
17 #include <memory>
18 #include <vector>
19 
20 #include "absl/types/optional.h"
21 #include "api/array_view.h"
22 #include "modules/audio_processing/agc/gain_control.h"
23 
24 namespace webrtc {
25 
26 class ApmDataDumper;
27 class AudioBuffer;
28 
29 class GainControlImpl : public GainControl {
30  public:
31   GainControlImpl();
32   GainControlImpl(const GainControlImpl&) = delete;
33   GainControlImpl& operator=(const GainControlImpl&) = delete;
34 
35   ~GainControlImpl() override;
36 
37   void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
38   int AnalyzeCaptureAudio(const AudioBuffer& audio);
39   int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
40 
41   void Initialize(size_t num_proc_channels, int sample_rate_hz);
42 
43   static void PackRenderAudioBuffer(const AudioBuffer& audio,
44                                     std::vector<int16_t>* packed_buffer);
45 
46   // GainControl implementation.
47   int stream_analog_level() const override;
is_limiter_enabled()48   bool is_limiter_enabled() const override { return limiter_enabled_; }
mode()49   Mode mode() const override { return mode_; }
50   int set_mode(Mode mode) override;
compression_gain_db()51   int compression_gain_db() const override { return compression_gain_db_; }
52   int set_analog_level_limits(int minimum, int maximum) override;
53   int set_compression_gain_db(int gain) override;
54   int set_target_level_dbfs(int level) override;
55   int enable_limiter(bool enable) override;
56   int set_stream_analog_level(int level) override;
57 
58  private:
59   struct MonoAgcState;
60 
61   // GainControl implementation.
target_level_dbfs()62   int target_level_dbfs() const override { return target_level_dbfs_; }
analog_level_minimum()63   int analog_level_minimum() const override { return minimum_capture_level_; }
analog_level_maximum()64   int analog_level_maximum() const override { return maximum_capture_level_; }
stream_is_saturated()65   bool stream_is_saturated() const override { return stream_is_saturated_; }
66 
67   int Configure();
68 
69   std::unique_ptr<ApmDataDumper> data_dumper_;
70 
71   const bool use_legacy_gain_applier_;
72   Mode mode_;
73   int minimum_capture_level_;
74   int maximum_capture_level_;
75   bool limiter_enabled_;
76   int target_level_dbfs_;
77   int compression_gain_db_;
78   int analog_capture_level_ = 0;
79   bool was_analog_level_set_;
80   bool stream_is_saturated_;
81 
82   std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
83   std::vector<int> capture_levels_;
84 
85   absl::optional<size_t> num_proc_channels_;
86   absl::optional<int> sample_rate_hz_;
87 
88   static int instance_counter_;
89 };
90 }  // namespace webrtc
91 
92 #endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
93