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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h | 67 |
1 files changed, 67 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h new file mode 100644 index 0000000000..3b43319951 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h @@ -0,0 +1,67 @@ +/* + * Copyright 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_AUDIO_PROCESSING_STATISTICS_H_ +#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ + +#include <stdint.h> + +#include "absl/types/optional.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { +// This version of the stats uses Optionals, it will replace the regular +// AudioProcessingStatistics struct. +struct RTC_EXPORT AudioProcessingStats { + AudioProcessingStats(); + AudioProcessingStats(const AudioProcessingStats& other); + ~AudioProcessingStats(); + + // Deprecated. + // TODO(bugs.webrtc.org/11226): Remove. + // True if voice is detected in the last capture frame, after processing. + // It is conservative in flagging audio as speech, with low likelihood of + // incorrectly flagging a frame as voice. + // Only reported if voice detection is enabled in AudioProcessing::Config. + absl::optional<bool> voice_detected; + + // AEC Statistics. + // ERL = 10log_10(P_far / P_echo) + absl::optional<double> echo_return_loss; + // ERLE = 10log_10(P_echo / P_out) + absl::optional<double> echo_return_loss_enhancement; + // Fraction of time that the AEC linear filter is divergent, in a 1-second + // non-overlapped aggregation window. + absl::optional<double> divergent_filter_fraction; + + // The delay metrics consists of the delay median and standard deviation. It + // also consists of the fraction of delay estimates that can make the echo + // cancellation perform poorly. The values are aggregated until the first + // call to `GetStatistics()` and afterwards aggregated and updated every + // second. Note that if there are several clients pulling metrics from + // `GetStatistics()` during a session the first call from any of them will + // change to one second aggregation window for all. + absl::optional<int32_t> delay_median_ms; + absl::optional<int32_t> delay_standard_deviation_ms; + + // Residual echo detector likelihood. + absl::optional<double> residual_echo_likelihood; + // Maximum residual echo likelihood from the last time period. + absl::optional<double> residual_echo_likelihood_recent_max; + + // The instantaneous delay estimate produced in the AEC. The unit is in + // milliseconds and the value is the instantaneous value at the time of the + // call to `GetStatistics()`. + absl::optional<int32_t> delay_ms; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_ |