tor-browser

The Tor Browser
git clone https://git.dasho.dev/tor-browser.git
Log | Files | Refs | README | LICENSE

gain_control_impl.h (2939B)

      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 <optional>
     19 #include <vector>
     20 
     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(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;
     48  bool is_limiter_enabled() const override { return limiter_enabled_; }
     49  Mode mode() const override { return mode_; }
     50  int set_mode(Mode mode) override;
     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.
     62  int target_level_dbfs() const override { return target_level_dbfs_; }
     63  int analog_level_minimum() const override { return minimum_capture_level_; }
     64  int analog_level_maximum() const override { return maximum_capture_level_; }
     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  Mode mode_;
     72  int minimum_capture_level_;
     73  int maximum_capture_level_;
     74  bool limiter_enabled_;
     75  int target_level_dbfs_;
     76  int compression_gain_db_;
     77  int analog_capture_level_ = 0;
     78  bool was_analog_level_set_;
     79  bool stream_is_saturated_;
     80 
     81  std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
     82  std::vector<int> capture_levels_;
     83 
     84  std::optional<size_t> num_proc_channels_;
     85  std::optional<int> sample_rate_hz_;
     86 
     87  static int instance_counter_;
     88 };
     89 }  // namespace webrtc
     90 
     91 #endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_