| /* |
| * 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 <stdint.h> |
| |
| #include <string> |
| |
| #include "modules/audio_processing/include/audio_frame_view.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "rtc_base/deprecation.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); |
| |
| 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; |
| int level; |
| bool keypress; |
| }; |
| |
| virtual ~AecDump() = default; |
| |
| // Logs Event::Type INIT message. |
| virtual void WriteInitMessage(const ProcessingConfig& api_format, |
| int64_t time_now_ms) = 0; |
| RTC_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<const float>& src) = 0; |
| virtual void AddCaptureStreamOutput( |
| const AudioFrameView<const float>& 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<const float>& 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_ |