#include <neteq.h>
◆ accelerate_rate
uint16_t webrtc::NetEqNetworkStatistics::accelerate_rate |
◆ added_zero_samples
size_t webrtc::NetEqNetworkStatistics::added_zero_samples |
◆ clockdrift_ppm
int32_t webrtc::NetEqNetworkStatistics::clockdrift_ppm |
◆ current_buffer_size_ms
uint16_t webrtc::NetEqNetworkStatistics::current_buffer_size_ms |
◆ expand_rate
uint16_t webrtc::NetEqNetworkStatistics::expand_rate |
◆ jitter_peaks_found
uint16_t webrtc::NetEqNetworkStatistics::jitter_peaks_found |
◆ max_waiting_time_ms
int webrtc::NetEqNetworkStatistics::max_waiting_time_ms |
◆ mean_waiting_time_ms
int webrtc::NetEqNetworkStatistics::mean_waiting_time_ms |
◆ median_waiting_time_ms
int webrtc::NetEqNetworkStatistics::median_waiting_time_ms |
◆ min_waiting_time_ms
int webrtc::NetEqNetworkStatistics::min_waiting_time_ms |
◆ packet_discard_rate
uint16_t webrtc::NetEqNetworkStatistics::packet_discard_rate |
◆ packet_loss_rate
uint16_t webrtc::NetEqNetworkStatistics::packet_loss_rate |
◆ preemptive_rate
uint16_t webrtc::NetEqNetworkStatistics::preemptive_rate |
◆ preferred_buffer_size_ms
uint16_t webrtc::NetEqNetworkStatistics::preferred_buffer_size_ms |
◆ secondary_decoded_rate
uint16_t webrtc::NetEqNetworkStatistics::secondary_decoded_rate |
◆ speech_expand_rate
uint16_t webrtc::NetEqNetworkStatistics::speech_expand_rate |
The documentation for this struct was generated from the following file:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/neteq/include/neteq.h