mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
109 lines
3.8 KiB
C++
109 lines
3.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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#include <stdint.h>
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#include <string>
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/deprecation.h"
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namespace webrtc {
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// Struct for passing current config from APM without having to
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// include protobuf headers.
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struct InternalAPMConfig {
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InternalAPMConfig();
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InternalAPMConfig(const InternalAPMConfig&);
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InternalAPMConfig(InternalAPMConfig&&);
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InternalAPMConfig& operator=(const InternalAPMConfig&);
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InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
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bool operator==(const InternalAPMConfig& other);
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bool aec_enabled = false;
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bool aec_delay_agnostic_enabled = false;
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bool aec_drift_compensation_enabled = false;
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bool aec_extended_filter_enabled = false;
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int aec_suppression_level = 0;
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bool aecm_enabled = false;
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bool aecm_comfort_noise_enabled = false;
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int aecm_routing_mode = 0;
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bool agc_enabled = false;
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int agc_mode = 0;
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bool agc_limiter_enabled = false;
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bool hpf_enabled = false;
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bool ns_enabled = false;
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int ns_level = 0;
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bool transient_suppression_enabled = false;
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bool noise_robust_agc_enabled = false;
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bool pre_amplifier_enabled = false;
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float pre_amplifier_fixed_gain_factor = 1.f;
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std::string experiments_description = "";
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};
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// An interface for recording configuration and input/output streams
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// of the Audio Processing Module. The recordings are called
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// 'aec-dumps' and are stored in a protobuf format defined in
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// debug.proto.
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// The Write* methods are always safe to call concurrently or
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// otherwise for all implementing subclasses. The intended mode of
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// operation is to create a protobuf object from the input, and send
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// it away to be written to file asynchronously.
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class AecDump {
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public:
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struct AudioProcessingState {
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int delay;
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int drift;
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int level;
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bool keypress;
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};
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virtual ~AecDump() = default;
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// Logs Event::Type INIT message.
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virtual void WriteInitMessage(const ProcessingConfig& api_format,
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int64_t time_now_ms) = 0;
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RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
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WriteInitMessage(api_format, 0);
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}
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// Logs Event::Type STREAM message. To log an input/output pair,
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// call the AddCapture* and AddAudioProcessingState methods followed
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// by a WriteCaptureStreamMessage call.
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virtual void AddCaptureStreamInput(
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const AudioFrameView<const float>& src) = 0;
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virtual void AddCaptureStreamOutput(
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const AudioFrameView<const float>& src) = 0;
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virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0;
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virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0;
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virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
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virtual void WriteCaptureStreamMessage() = 0;
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// Logs Event::Type REVERSE_STREAM message.
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virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
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virtual void WriteRenderStreamMessage(
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const AudioFrameView<const float>& src) = 0;
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virtual void WriteRuntimeSetting(
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const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
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// Logs Event::Type CONFIG message.
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virtual void WriteConfig(const InternalAPMConfig& config) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
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