NetEq/ACM: Refactor how packet waiting times are calculated

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
 #include <algorithm>  // sort
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/main/acm2/nack.h"
 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
@@ skipping 612 matching lines @@
   acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
   acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
   acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
   acm_stat->currentExpandRate = neteq_stat.expand_rate;
   acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
   acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
   acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
   acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
   acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
   acm_stat->addedSamples = neteq_stat.added_zero_samples;
-
-  std::vector<int> waiting_times;
-  neteq_->WaitingTimes(&waiting_times);
-  size_t size = waiting_times.size();
-  if (size == 0) {
-    acm_stat->meanWaitingTimeMs = -1;
-    acm_stat->medianWaitingTimeMs = -1;
-    acm_stat->minWaitingTimeMs = -1;
-    acm_stat->maxWaitingTimeMs = -1;
-  } else {
-    std::sort(waiting_times.begin(), waiting_times.end());
-    if ((size & 0x1) == 0) {
-      acm_stat->medianWaitingTimeMs = (waiting_times[size / 2 - 1] +
-          waiting_times[size / 2]) / 2;
-    } else {
-      acm_stat->medianWaitingTimeMs = waiting_times[size / 2];
-    }
-    acm_stat->minWaitingTimeMs = waiting_times.front();
-    acm_stat->maxWaitingTimeMs = waiting_times.back();
-    double sum = 0;
-    for (size_t i = 0; i < size; ++i) {
-      sum += waiting_times[i];
-    }
-    acm_stat->meanWaitingTimeMs = static_cast<int>(sum / size);
-  }
+  acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
+  acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
+  acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
+  acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
 }
 
 int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
                                       CodecInst* codec) const {
   CriticalSectionScoped lock(crit_sect_.get());
   auto it = decoders_.find(payload_type);
   if (it == decoders_.end()) {
     LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
                 << static_cast<int>(payload_type);
     return -1;
@@ skipping 151 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc