#include <audio_processing_simulator.h>
◆ SimulationSettings() [1/4]
webrtc::test::SimulationSettings::SimulationSettings |
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◆ SimulationSettings() [2/4]
◆ ~SimulationSettings() [1/2]
webrtc::test::SimulationSettings::~SimulationSettings |
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◆ SimulationSettings() [3/4]
webrtc::test::SimulationSettings::SimulationSettings |
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◆ SimulationSettings() [4/4]
◆ ~SimulationSettings() [2/2]
webrtc::test::SimulationSettings::~SimulationSettings |
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◆ aec_dump_input_filename
◆ aec_dump_output_filename
◆ aec_suppression_level
rtc::Optional< int > webrtc::test::SimulationSettings::aec_suppression_level |
◆ aecm_routing_mode
rtc::Optional< int > webrtc::test::SimulationSettings::aecm_routing_mode |
◆ agc_compression_gain
rtc::Optional< int > webrtc::test::SimulationSettings::agc_compression_gain |
◆ agc_mode
◆ agc_target_level
rtc::Optional< int > webrtc::test::SimulationSettings::agc_target_level |
◆ artificial_nearend_filename
◆ custom_call_order_filename
◆ discard_all_settings_in_aecdump
bool webrtc::test::SimulationSettings::discard_all_settings_in_aecdump = true |
◆ ed_graph_output_filename
◆ fixed_interface
bool webrtc::test::SimulationSettings::fixed_interface = false |
◆ input_filename
◆ microphone_positions
◆ ns_level
◆ output_filename
◆ output_num_channels
rtc::Optional< int > webrtc::test::SimulationSettings::output_num_channels |
◆ output_sample_rate_hz
rtc::Optional< int > webrtc::test::SimulationSettings::output_sample_rate_hz |
◆ report_bitexactness
bool webrtc::test::SimulationSettings::report_bitexactness = false |
◆ report_performance
bool webrtc::test::SimulationSettings::report_performance = false |
◆ reverse_input_filename
◆ reverse_output_filename
◆ reverse_output_num_channels
rtc::Optional< int > webrtc::test::SimulationSettings::reverse_output_num_channels |
◆ reverse_output_sample_rate_hz
rtc::Optional< int > webrtc::test::SimulationSettings::reverse_output_sample_rate_hz |
◆ store_intermediate_output
bool webrtc::test::SimulationSettings::store_intermediate_output = false |
◆ stream_delay
rtc::Optional< int > webrtc::test::SimulationSettings::stream_delay |
◆ stream_drift_samples
rtc::Optional< int > webrtc::test::SimulationSettings::stream_drift_samples |
◆ target_angle_degrees
int webrtc::test::SimulationSettings::target_angle_degrees = 90 |
◆ use_aec
◆ use_aec3
◆ use_aecm
◆ use_aecm_comfort_noise
rtc::Optional< bool > webrtc::test::SimulationSettings::use_aecm_comfort_noise |
◆ use_agc
◆ use_agc_limiter
rtc::Optional< bool > webrtc::test::SimulationSettings::use_agc_limiter |
◆ use_all
◆ use_bf
◆ use_delay_agnostic
rtc::Optional< bool > webrtc::test::SimulationSettings::use_delay_agnostic |
◆ use_drift_compensation
rtc::Optional< bool > webrtc::test::SimulationSettings::use_drift_compensation |
◆ use_ed
◆ use_experimental_agc
rtc::Optional< bool > webrtc::test::SimulationSettings::use_experimental_agc |
◆ use_extended_filter
rtc::Optional< bool > webrtc::test::SimulationSettings::use_extended_filter |
◆ use_hpf
◆ use_ie
◆ use_lc
◆ use_le
◆ use_ns
◆ use_refined_adaptive_filter
rtc::Optional< bool > webrtc::test::SimulationSettings::use_refined_adaptive_filter |
◆ use_ts
◆ use_vad
◆ use_verbose_logging
bool webrtc::test::SimulationSettings::use_verbose_logging = false |
◆ vad_likelihood
rtc::Optional< int > webrtc::test::SimulationSettings::vad_likelihood |
The documentation for this struct was generated from the following files: