| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
| #define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include <memory> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "modules/audio_processing/agc/gain_control.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| class AudioBuffer; |
| |
| class GainControlImpl : public GainControl { |
| public: |
| GainControlImpl(); |
| GainControlImpl(const GainControlImpl&) = delete; |
| GainControlImpl& operator=(const GainControlImpl&) = delete; |
| |
| ~GainControlImpl() override; |
| |
| void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio); |
| int AnalyzeCaptureAudio(const AudioBuffer& audio); |
| int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo); |
| |
| void Initialize(size_t num_proc_channels, int sample_rate_hz); |
| |
| static void PackRenderAudioBuffer(const AudioBuffer& audio, |
| std::vector<int16_t>* packed_buffer); |
| |
| // GainControl implementation. |
| bool is_enabled() const override { return enabled_; } |
| int stream_analog_level() const override; |
| bool is_limiter_enabled() const override { return limiter_enabled_; } |
| Mode mode() const override { return mode_; } |
| int Enable(bool enable) override; |
| int set_mode(Mode mode) override; |
| int compression_gain_db() const override { return compression_gain_db_; } |
| int set_analog_level_limits(int minimum, int maximum) override; |
| int set_compression_gain_db(int gain) override; |
| int set_target_level_dbfs(int level) override; |
| int enable_limiter(bool enable) override; |
| int set_stream_analog_level(int level) override; |
| |
| private: |
| struct MonoAgcState; |
| |
| // GainControl implementation. |
| int target_level_dbfs() const override { return target_level_dbfs_; } |
| int analog_level_minimum() const override { return minimum_capture_level_; } |
| int analog_level_maximum() const override { return maximum_capture_level_; } |
| bool stream_is_saturated() const override { return stream_is_saturated_; } |
| |
| int Configure(); |
| |
| std::unique_ptr<ApmDataDumper> data_dumper_; |
| |
| bool enabled_ = false; |
| |
| const bool use_legacy_gain_applier_; |
| Mode mode_; |
| int minimum_capture_level_; |
| int maximum_capture_level_; |
| bool limiter_enabled_; |
| int target_level_dbfs_; |
| int compression_gain_db_; |
| int analog_capture_level_; |
| bool was_analog_level_set_; |
| bool stream_is_saturated_; |
| |
| std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_; |
| std::vector<int> capture_levels_; |
| |
| absl::optional<size_t> num_proc_channels_; |
| absl::optional<int> sample_rate_hz_; |
| |
| static int instance_counter_; |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |