AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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16
webrtc/modules/audio_processing/include/audio_processing_statistics.h
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@ -24,14 +24,8 @@ struct RTC_EXPORT AudioProcessingStats {
AudioProcessingStats(const AudioProcessingStats& other);
~AudioProcessingStats();
// The root mean square (RMS) level in dBFS (decibels from digital
// full-scale) of the last capture frame, after processing. It is
// constrained to [-127, 0].
// The computation follows: https://tools.ietf.org/html/rfc6465
// with the intent that it can provide the RTP audio level indication.
// Only reported if level estimation is enabled in AudioProcessing::Config.
absl::optional<int> output_rms_dbfs;
// 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.
@ -50,9 +44,9 @@ struct RTC_EXPORT AudioProcessingStats {
// 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
// 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
// `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;
@ -64,7 +58,7 @@ struct RTC_EXPORT AudioProcessingStats {
// 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()|.
// call to `GetStatistics()`.
absl::optional<int32_t> delay_ms;
};