Tolerate out of order samples to SendProcessingUsage2::FrameSent.

Bug: chromium:842613
Change-Id: I57e4df75dcfdfb9bf42819f31d2186e875a90a3a
Reviewed-on: https://webrtc-review.googlesource.com/92880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24224}

video/overuse_frame_detector.cc

@@ -249,6 +249,14 @@ class SendProcessingUsage2 : public OveruseFrameDetector::ProcessingUsage {
       absl::optional<int> encode_duration_us) override {
     if (encode_duration_us) {
       if (prev_time_us_ != -1) {
+        if (capture_time_us < prev_time_us_) {
+          // The weighting in AddSample assumes that samples are processed with
+          // non-decreasing measurement timestamps. We could implement
+          // appropriate weights for samples arriving late, but since it is a
+          // rare case, keep things simple, by just pushing those measurements a
+          // bit forward in time.
+          capture_time_us = prev_time_us_;
+        }
         AddSample(1e-6 * (*encode_duration_us),
                   1e-6 * (capture_time_us - prev_time_us_));
       }

video/overuse_frame_detector_unittest.cc

@@ -890,4 +890,19 @@ TEST_F(OveruseFrameDetectorTest2, NoOveruseForRandomFrameIntervalWithReset) {
   EXPECT_LE(UsagePercent(), InitialUsage() + 5);
 }
 
+TEST_F(OveruseFrameDetectorTest2, ToleratesOutOfOrderFrames) {
+  overuse_detector_->SetOptions(options_);
+  // Represents a cpu utilization close to 100%. First input frame results in
+  // three encoded frames, and the last of those isn't finished until after the
+  // first encoded frame corresponding to the next input frame.
+  const int kEncodeTimeUs = 30 * rtc::kNumMicrosecsPerMillisec;
+  const int kCaptureTimesMs[] = { 33, 33, 66, 33 };
+
+  for (int capture_time_ms : kCaptureTimesMs) {
+    overuse_detector_->FrameSent(
+        0, 0, capture_time_ms * rtc::kNumMicrosecsPerMillisec, kEncodeTimeUs);
+  }
+  EXPECT_GE(UsagePercent(), InitialUsage());
+}
+
 }  // namespace webrtc