Adds UMA histogram for system delay jumps

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.
  */
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
 #include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+}
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/file_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/audio_processing/debug.pb.h"
 #endif
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
@@ skipping 117 matching lines @@
 #endif
       fwd_in_format_(kSampleRate16kHz, 1),
       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),
       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 397 matching lines @@
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
     audioproc::Stream* msg = event_msg_->mutable_stream();
     msg->set_delay(stream_delay_ms_);
     msg->set_drift(echo_cancellation_->stream_drift_samples());
     msg->set_level(gain_control()->stream_analog_level());
     msg->set_keypress(key_pressed_);
   }
 #endif
 
+  MaybeUpdateHistograms();
+
   AudioBuffer* ca = capture_audio_.get();  // For brevity.
   if (use_new_agc_ && gain_control_->is_enabled()) {
     agc_manager_->AnalyzePreProcess(ca->channels()[0],
                                     ca->num_channels(),
                                     fwd_proc_format_.samples_per_channel());
   }
 
   bool data_processed = is_data_processed();
   if (analysis_needed(data_processed)) {
     ca->SplitIntoFrequencyBands();
@@ skipping 383 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;
+
+  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;
+  }
+}
+
 #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