f3aa6326b8
This CL replaces the use of the ExperimentalAgc config with using the new config format. Beyond that, some further changes were made to how the analog and digital AGCs are initialized/called. While these can be made in a separate CL, I believe the code changes becomes more clear by bundling those with the replacement of the ExperimentalAgc config. TBR: saza@webrtc.org Bug: webrtc:5298 Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30149}
93 lines
2.9 KiB
C++
93 lines
2.9 KiB
C++
/*
|
|
* 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.
|
|
int stream_analog_level() const override;
|
|
bool is_limiter_enabled() const override { return limiter_enabled_; }
|
|
Mode mode() const override { return mode_; }
|
|
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_;
|
|
|
|
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_ = 0;
|
|
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_
|