Use the new AudioProcessing statistics everywhere.
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}diff --git a/modules/audio_processing/include/audio_processing_statistics.h b/modules/audio_processing/include/audio_processing_statistics.h
new file mode 100644
index 0000000..7dbc907
--- /dev/null
+++ b/modules/audio_processing/include/audio_processing_statistics.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2017 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_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+
+#include "api/optional.h"
+
+namespace webrtc {
+// This version of the stats uses Optionals, it will replace the regular
+// AudioProcessingStatistics struct.
+struct AudioProcessingStats {
+ AudioProcessingStats();
+ AudioProcessingStats(const AudioProcessingStats& other);
+ ~AudioProcessingStats();
+
+ // AEC Statistics.
+ // ERL = 10log_10(P_far / P_echo)
+ rtc::Optional<double> echo_return_loss;
+ // ERLE = 10log_10(P_echo / P_out)
+ rtc::Optional<double> echo_return_loss_enhancement;
+ // Fraction of time that the AEC linear filter is divergent, in a 1-second
+ // non-overlapped aggregation window.
+ rtc::Optional<double> divergent_filter_fraction;
+
+ // 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
+ // second. Note that if there are several clients pulling metrics from
+ // |GetStatistics()| during a session the first call from any of them will
+ // change to one second aggregation window for all.
+ rtc::Optional<int32_t> delay_median_ms;
+ rtc::Optional<int32_t> delay_standard_deviation_ms;
+
+ // Residual echo detector likelihood.
+ rtc::Optional<double> residual_echo_likelihood;
+ // Maximum residual echo likelihood from the last time period.
+ rtc::Optional<double> residual_echo_likelihood_recent_max;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_