/* * Copyright (c) 2017 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_INCLUDE_AEC_DUMP_H_ #define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_ #include #include #include "absl/base/attributes.h" #include "absl/types/optional.h" #include "modules/audio_processing/include/audio_frame_view.h" #include "modules/audio_processing/include/audio_processing.h" namespace webrtc { // Struct for passing current config from APM without having to // include protobuf headers. struct InternalAPMConfig { InternalAPMConfig(); InternalAPMConfig(const InternalAPMConfig&); InternalAPMConfig(InternalAPMConfig&&); InternalAPMConfig& operator=(const InternalAPMConfig&); InternalAPMConfig& operator=(InternalAPMConfig&&) = delete; bool operator==(const InternalAPMConfig& other) const; bool aec_enabled = false; bool aec_delay_agnostic_enabled = false; bool aec_drift_compensation_enabled = false; bool aec_extended_filter_enabled = false; int aec_suppression_level = 0; bool aecm_enabled = false; bool aecm_comfort_noise_enabled = false; int aecm_routing_mode = 0; bool agc_enabled = false; int agc_mode = 0; bool agc_limiter_enabled = false; bool hpf_enabled = false; bool ns_enabled = false; int ns_level = 0; bool transient_suppression_enabled = false; bool noise_robust_agc_enabled = false; bool pre_amplifier_enabled = false; float pre_amplifier_fixed_gain_factor = 1.f; std::string experiments_description = ""; }; // An interface for recording configuration and input/output streams // of the Audio Processing Module. The recordings are called // 'aec-dumps' and are stored in a protobuf format defined in // debug.proto. // The Write* methods are always safe to call concurrently or // otherwise for all implementing subclasses. The intended mode of // operation is to create a protobuf object from the input, and send // it away to be written to file asynchronously. class AecDump { public: struct AudioProcessingState { int delay; int drift; absl::optional applied_input_volume; bool keypress; }; virtual ~AecDump() = default; // Logs Event::Type INIT message. virtual void WriteInitMessage(const ProcessingConfig& api_format, int64_t time_now_ms) = 0; ABSL_DEPRECATED("") void WriteInitMessage(const ProcessingConfig& api_format) { WriteInitMessage(api_format, 0); } // Logs Event::Type STREAM message. To log an input/output pair, // call the AddCapture* and AddAudioProcessingState methods followed // by a WriteCaptureStreamMessage call. virtual void AddCaptureStreamInput( const AudioFrameView& src) = 0; virtual void AddCaptureStreamOutput( const AudioFrameView& src) = 0; virtual void AddCaptureStreamInput(const int16_t* const data, int num_channels, int samples_per_channel) = 0; virtual void AddCaptureStreamOutput(const int16_t* const data, int num_channels, int samples_per_channel) = 0; virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0; virtual void WriteCaptureStreamMessage() = 0; // Logs Event::Type REVERSE_STREAM message. virtual void WriteRenderStreamMessage(const int16_t* const data, int num_channels, int samples_per_channel) = 0; virtual void WriteRenderStreamMessage( const AudioFrameView& src) = 0; virtual void WriteRuntimeSetting( const AudioProcessing::RuntimeSetting& runtime_setting) = 0; // Logs Event::Type CONFIG message. virtual void WriteConfig(const InternalAPMConfig& config) = 0; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_