gain_control_impl.h 2.9 KB

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