Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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_