audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 982 matching lines @@
 void AudioProcessingImpl::InitializeBeamformer() {
   if (beamformer_enabled_) {
     if (!beamformer_) {
       beamformer_.reset(new NonlinearBeamformer(array_geometry_));
     }
     beamformer_->Initialize(kChunkSizeMs, split_rate_);
   }
 }
 
 void AudioProcessingImpl::MaybeUpdateHistograms() {
-  static const int kMinDiffDelayMs = 50;
+  static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
     // Detect a jump in platform reported system delay and log the difference.
     const int diff_stream_delay_ms = stream_delay_ms_ - last_stream_delay_ms_;
     if (diff_stream_delay_ms > kMinDiffDelayMs && last_stream_delay_ms_ != 0) {
       RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
                            diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
     }
     last_stream_delay_ms_ = stream_delay_ms_;
 
     // Detect a jump in AEC system delay and log the difference.
     const int frames_per_ms = rtc::CheckedDivExact(split_rate_, 1000);
     const int aec_system_delay_ms =
         WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
     const int diff_aec_system_delay_ms = aec_system_delay_ms -
         last_aec_system_delay_ms_;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
         last_aec_system_delay_ms_ != 0) {
       RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
                            diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
                            100);
     }
     last_aec_system_delay_ms_ = aec_system_delay_ms;
+    // TODO(bjornv): Consider also logging amount of jumps. This gives a better
+    // indication of how frequent jumps are.
   }
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile() {
   int32_t size = event_msg_->ByteSize();
   if (size <= 0) {
     return kUnspecifiedError;
   }
 #if defined(WEBRTC_ARCH_BIG_ENDIAN)
@@ skipping 31 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc