AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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55
webrtc/modules/audio_processing/include/audio_processing_statistics.h
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@ -11,57 +11,8 @@
#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
// This is a transitional header forwarding to the new version in the api/
// folder.
#include "api/audio/audio_processing_statistics.h"
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_