Replacing NetEq discard rate with secondary discarded rate.

NetEq network statistics contains discard rate but has not been used and even not been implemented until recently.

According to w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded,
this statistics needs to be replaced with an accumulative stats. Such work will be carried out separately.

Meanwhile, we need to add a rate to reflect rate of discarded redundant packets. See webrtc:8025.

In this CL, we replace the existing discard rate with secondary discarded rate, so as to
1. fulfill the requests on webrtc:8025
2. get ready to implement an accumulative statistics for discarded packets.

BUG: webrtc:7903,webrtc:8025
Change-Id: Idbf143a105db76ca15f0af54848e1448f2a810ec
Reviewed-on: https://chromium-review.googlesource.com/582863
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19495}

webrtc/modules/audio_coding/neteq/statistics_calculator.cc

diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index 145d99a..3faed62 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -123,14 +123,14 @@
       lost_timestamps_(0),
       timestamps_since_last_report_(0),
       secondary_decoded_samples_(0),
+      discarded_secondary_packets_(0),
       delayed_packet_outage_counter_(
           "WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
           60000,  // 60 seconds report interval.
           100),
       excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
                            60000,  // 60 seconds report interval.
-                           1000) {
-}
+                           1000) {}
 
 StatisticsCalculator::~StatisticsCalculator() = default;
 
@@ -141,6 +141,7 @@
   expanded_speech_samples_ = 0;
   expanded_noise_samples_ = 0;
   secondary_decoded_samples_ = 0;
+  discarded_secondary_packets_ = 0;
   waiting_times_.clear();
 }
 
@@ -184,6 +185,10 @@
   discarded_packets_ += num_packets;
 }
 
+void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
+  discarded_secondary_packets_ += num_packets;
+}
+
 void StatisticsCalculator::LostSamples(size_t num_samples) {
   lost_timestamps_ += num_samples;
 }
@@ -249,10 +254,6 @@
   stats->packet_loss_rate =
       CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
 
-  const size_t discarded_samples = discarded_packets_ * samples_per_packet;
-  stats->packet_discard_rate =
-      CalculateQ14Ratio(discarded_samples, timestamps_since_last_report_);
-
   stats->accelerate_rate =
       CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);
 
@@ -271,6 +272,12 @@
       CalculateQ14Ratio(secondary_decoded_samples_,
                         timestamps_since_last_report_);
 
+  const size_t discarded_secondary_samples =
+      discarded_secondary_packets_ * samples_per_packet;
+  stats->secondary_discarded_rate = CalculateQ14Ratio(
+      discarded_secondary_samples,
+      discarded_secondary_samples + secondary_decoded_samples_);
+
   if (waiting_times_.size() == 0) {
     stats->mean_waiting_time_ms = -1;
     stats->median_waiting_time_ms = -1;