mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
96 lines
2.8 KiB
C++
96 lines
2.8 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/include/gain_control.h"
|
|
#include "rtc_base/constructor_magic.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(AudioBuffer* audio);
|
|
int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
|
|
|
|
void Initialize(size_t num_proc_channels, int sample_rate_hz);
|
|
|
|
static void PackRenderAudioBuffer(AudioBuffer* audio,
|
|
std::vector<int16_t>* packed_buffer);
|
|
|
|
// GainControl implementation.
|
|
bool is_enabled() const override;
|
|
int stream_analog_level() const override;
|
|
bool is_limiter_enabled() const override;
|
|
Mode mode() const override;
|
|
|
|
int compression_gain_db() const override;
|
|
|
|
private:
|
|
class GainController;
|
|
|
|
// GainControl implementation.
|
|
int Enable(bool enable) override;
|
|
int set_stream_analog_level(int level) override;
|
|
int set_mode(Mode mode) override;
|
|
int set_target_level_dbfs(int level) override;
|
|
int target_level_dbfs() const override;
|
|
int set_compression_gain_db(int gain) override;
|
|
int enable_limiter(bool enable) override;
|
|
int set_analog_level_limits(int minimum, int maximum) override;
|
|
int analog_level_minimum() const override;
|
|
int analog_level_maximum() const override;
|
|
bool stream_is_saturated() const override;
|
|
|
|
int Configure();
|
|
|
|
std::unique_ptr<ApmDataDumper> data_dumper_;
|
|
|
|
bool enabled_ = false;
|
|
|
|
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<GainController>> gain_controllers_;
|
|
|
|
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_
|