audio_processing/aec: Increased delay metrics aggregation window to five seconds
The known clients (GetStats and UMA histogram in Chrome) use at least 5 second aggregation window. There is no particular value in calculating the metrics more often. The CL also includes a small refactoring moving a declaration inside an if statement. BUG=2994 TEST=N/A R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40219004 Cr-Commit-Position: refs/heads/master@{#8619} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8619 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -39,7 +39,7 @@ static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
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// Metrics
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static const int subCountLen = 4;
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static const int countLen = 50;
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static const int kDelayMetricsAggregationWindow = 250; // One second at 16 kHz.
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static const int kDelayMetricsAggregationWindow = 1250; // 5 seconds at 16 kHz.
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// Quantities to control H band scaling for SWB input
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static const int flagHbandCn = 1; // flag for adding comfort noise in H band
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@ -1249,10 +1249,9 @@ static void ProcessBlock(AecCore* aec) {
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// Block wise delay estimation used for logging
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if (aec->delay_logging_enabled) {
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int delay_estimate = 0;
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if (WebRtc_AddFarSpectrumFloat(
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aec->delay_estimator_farend, abs_far_spectrum, PART_LEN1) == 0) {
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delay_estimate = WebRtc_DelayEstimatorProcessFloat(
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int delay_estimate = WebRtc_DelayEstimatorProcessFloat(
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aec->delay_estimator, abs_near_spectrum, PART_LEN1);
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if (delay_estimate >= 0) {
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// Update delay estimate buffer.
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