#include <mediastreaminterface.h>
◆ AudioProcessorStats() [1/2]
| webrtc::AudioProcessorInterface::AudioProcessorStats::AudioProcessorStats |
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◆ ~AudioProcessorStats() [1/2]
| webrtc::AudioProcessorInterface::AudioProcessorStats::~AudioProcessorStats |
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◆ AudioProcessorStats() [2/2]
| webrtc::AudioProcessorInterface::AudioProcessorStats::AudioProcessorStats |
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◆ ~AudioProcessorStats() [2/2]
| webrtc::AudioProcessorInterface::AudioProcessorStats::~AudioProcessorStats |
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◆ aec_divergent_filter_fraction
| float webrtc::AudioProcessorInterface::AudioProcessorStats::aec_divergent_filter_fraction |
◆ aec_quality_min
| float webrtc::AudioProcessorInterface::AudioProcessorStats::aec_quality_min |
◆ echo_delay_median_ms
| int webrtc::AudioProcessorInterface::AudioProcessorStats::echo_delay_median_ms |
◆ echo_delay_std_ms
| int webrtc::AudioProcessorInterface::AudioProcessorStats::echo_delay_std_ms |
◆ echo_return_loss
| int webrtc::AudioProcessorInterface::AudioProcessorStats::echo_return_loss |
◆ echo_return_loss_enhancement
| int webrtc::AudioProcessorInterface::AudioProcessorStats::echo_return_loss_enhancement |
◆ residual_echo_likelihood
| float webrtc::AudioProcessorInterface::AudioProcessorStats::residual_echo_likelihood |
◆ residual_echo_likelihood_recent_max
| float webrtc::AudioProcessorInterface::AudioProcessorStats::residual_echo_likelihood_recent_max |
◆ typing_noise_detected
| bool webrtc::AudioProcessorInterface::AudioProcessorStats::typing_noise_detected |
The documentation for this struct was generated from the following file: