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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h | 60 |
1 files changed, 60 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h new file mode 100644 index 0000000000..dd1fa82e93 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2018 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_AEC3_API_CALL_JITTER_METRICS_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_ + +namespace webrtc { + +// Stores data for reporting metrics on the API call jitter. +class ApiCallJitterMetrics { + public: + class Jitter { + public: + Jitter(); + void Update(int num_api_calls_in_a_row); + void Reset(); + + int min() const { return min_; } + int max() const { return max_; } + + private: + int max_; + int min_; + }; + + ApiCallJitterMetrics() { Reset(); } + + // Update metrics for render API call. + void ReportRenderCall(); + + // Update and periodically report metrics for capture API call. + void ReportCaptureCall(); + + // Methods used only for testing. + const Jitter& render_jitter() const { return render_jitter_; } + const Jitter& capture_jitter() const { return capture_jitter_; } + bool WillReportMetricsAtNextCapture() const; + + private: + void Reset(); + + Jitter render_jitter_; + Jitter capture_jitter_; + + int num_api_calls_in_a_row_ = 0; + int frames_since_last_report_ = 0; + bool last_call_was_render_ = false; + bool proper_call_observed_ = false; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_ |