audio_processing: Adds two UMA histograms logging delay jumps in AEC

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 159 matching lines @@
       fwd_proc_format_(kSampleRate16kHz),
       fwd_out_format_(kSampleRate16kHz, 1),
       rev_in_format_(kSampleRate16kHz, 1),
       rev_proc_format_(kSampleRate16kHz, 1),
       split_rate_(kSampleRate16kHz),
       stream_delay_ms_(0),
       delay_offset_ms_(0),
       was_stream_delay_set_(false),
       last_stream_delay_ms_(0),
       last_aec_system_delay_ms_(0),
+      stream_delay_jumps_(-1),
+      aec_system_delay_jumps_(-1),
       output_will_be_muted_(false),
       key_pressed_(false),
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       use_new_agc_(false),
 #else
       use_new_agc_(config.Get<ExperimentalAgc>().enabled),
 #endif
       agc_startup_min_volume_(config.Get<ExperimentalAgc>().startup_min_volume),
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       transient_suppressor_enabled_(false),
@@ skipping 806 matching lines @@
       beamformer_.reset(new NonlinearBeamformer(array_geometry_));
     }
     beamformer_->Initialize(kChunkSizeMs, split_rate_);
   }
 }
 
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
+    // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
+    // If a stream has echo we know that the echo_cancellation is in process.
+    if (stream_delay_jumps_ == -1 && echo_cancellation()->stream_has_echo()) {
+      stream_delay_jumps_ = 0;
+    }
+    if (aec_system_delay_jumps_ == -1 &&
+        echo_cancellation()->stream_has_echo()) {
+      aec_system_delay_jumps_ = 0;
+    }
+
     // 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);
+      if (stream_delay_jumps_ == -1) {
+        stream_delay_jumps_ = 0;  // Activate counter if needed.
+      }
+      stream_delay_jumps_++;
     }
     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);
+      if (aec_system_delay_jumps_ == -1) {
+        aec_system_delay_jumps_ = 0;  // Activate counter if needed.
+      }
+      aec_system_delay_jumps_++;
     }
     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.
   }
 }
 
+void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
+  CriticalSectionScoped crit_scoped(crit_);
+  if (stream_delay_jumps_ > -1) {
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
+        stream_delay_jumps_, 51);
+  }
+  stream_delay_jumps_ = -1;
+  last_stream_delay_ms_ = 0;
+
+  if (aec_system_delay_jumps_ > -1) {
+    RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
+                              aec_system_delay_jumps_, 51);
+  }
+  aec_system_delay_jumps_ = -1;
+  last_aec_system_delay_ms_ = 0;
+}
+
 #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)
   // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
   //            pretty safe in assuming little-endian.
 #endif
@@ skipping 28 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc