mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00
2bfa767245
Pass the correct number of channels needed by the AGC2 input volume controller. This change doesn't affect the adaptive digital controller which reads the number of channel from the passed audio buffer instance for each processed frame. Note that the `AdaptiveDigitalGainController::Initialize()` impl was removed in [1], but that CL didn't remove the declaration (done in this CL). [1] https://webrtc-review.googlesource.com/c/src/+/287222/5/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc#105 Bug: webrtc:7494 Change-Id: I07369ab4025a251b25c716cf618e4222fdb60fc8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287320 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38863}
66 lines
2.5 KiB
C++
66 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2018 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_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "modules/audio_processing/agc2/gain_applier.h"
|
|
#include "modules/audio_processing/include/audio_frame_view.h"
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class ApmDataDumper;
|
|
|
|
// Selects the target digital gain, decides when and how quickly to adapt to the
|
|
// target and applies the current gain to 10 ms frames.
|
|
class AdaptiveDigitalGainController {
|
|
public:
|
|
// Information about a frame to process.
|
|
struct FrameInfo {
|
|
float speech_probability; // Probability of speech in the [0, 1] range.
|
|
float speech_level_dbfs; // Estimated speech level (dBFS).
|
|
bool speech_level_reliable; // True with reliable speech level estimation.
|
|
float noise_rms_dbfs; // Estimated noise RMS level (dBFS).
|
|
float headroom_db; // Headroom (dB).
|
|
// TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
|
|
float limiter_envelope_dbfs; // Envelope level from the limiter (dBFS).
|
|
};
|
|
|
|
AdaptiveDigitalGainController(
|
|
ApmDataDumper* apm_data_dumper,
|
|
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
|
|
int adjacent_speech_frames_threshold);
|
|
AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
|
|
AdaptiveDigitalGainController& operator=(
|
|
const AdaptiveDigitalGainController&) = delete;
|
|
|
|
// Analyzes `info`, updates the digital gain and applies it to a 10 ms
|
|
// `frame`. Supports any sample rate supported by APM.
|
|
void Process(const FrameInfo& info, AudioFrameView<float> frame);
|
|
|
|
private:
|
|
ApmDataDumper* const apm_data_dumper_;
|
|
GainApplier gain_applier_;
|
|
|
|
const AudioProcessing::Config::GainController2::AdaptiveDigital config_;
|
|
const int adjacent_speech_frames_threshold_;
|
|
const float max_gain_change_db_per_10ms_;
|
|
|
|
int calls_since_last_gain_log_;
|
|
int frames_to_gain_increase_allowed_;
|
|
float last_gain_db_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
|