UMA stats for browser/renderer audio rendering buffer size mismatch.

media/audio/audio_output_resampler.cc

Index: media/audio/audio_output_resampler.cc
diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc
index 54afbe99fde86847835977ddbc4cebd6180c8013..4c2e6e2b68618322a1f7064a6b7fb22b422400ea 100644
--- a/media/audio/audio_output_resampler.cc
+++ b/media/audio/audio_output_resampler.cc
@@ -11,6 +11,7 @@
 #include "base/compiler_specific.h"
 #include "base/macros.h"
 #include "base/metrics/histogram.h"
+#include "base/metrics/sparse_histogram.h"
 #include "base/numerics/safe_conversions.h"
 #include "base/single_thread_task_runner.h"
 #include "base/trace_event/trace_event.h"
@@ -135,6 +136,39 @@ static void RecordFallbackStats(const AudioParameters& output_params) {
   }
 }
 
+// Record UMA statistics for input/output rebuffering.
+static void RecordRebufferingStats(const AudioParameters& input_params,
+                                   const AudioParameters& output_params) {
+  const int& input_buffer_size = input_params.frames_per_buffer();
+  const int& output_buffer_size = output_params.frames_per_buffer();
+  DCHECK(input_buffer_size);
+  DCHECK(output_buffer_size);
+  int value = (output_buffer_size - input_buffer_size) * 1000 /
+              std::max(input_buffer_size, output_buffer_size);

[Henrik Grunell, 2016/08/23 15:53:19]

I believe you had it the other way around (i.e. min(i, o)) when you presented
the idea to me. Maybe you wanted to switch which case should be negative?

[o1ka, 2016/08/23 16:20:56]

Right, it's a misprint, thanks for catching!

+
+  switch (input_params.latency_tag()) {
+    case AudioLatency::LATENCY_EXACT_MS:
+      UMA_HISTOGRAM_SPARSE_SLOWLY(
+          "Media.Audio.Render.BufferSizeMismatch.LatencyExactMs", value);
+      return;
+    case AudioLatency::LATENCY_INTERACTIVE:
+      UMA_HISTOGRAM_SPARSE_SLOWLY(
+          "Media.Audio.Render.BufferSizeMismatch.LatencyInteractive", value);
+      return;
+    case AudioLatency::LATENCY_RTC:
+      UMA_HISTOGRAM_SPARSE_SLOWLY(
+          "Media.Audio.Render.BufferSizeMismatch.LatencyRtc", value);
+      return;
+    case AudioLatency::LATENCY_PLAYBACK:
+      UMA_HISTOGRAM_SPARSE_SLOWLY(
+          "Media.Audio.Render.BufferSizeMismatch.LatencyPlayback", value);
+      return;
+    default:
+      UMA_HISTOGRAM_SPARSE_SLOWLY(

[Henrik Grunell, 2016/08/23 15:53:19]

This should never happen, right? Maybe NOTREACHED() and no histogram?

[o1ka, 2016/08/23 16:20:56]

Why not? It can be the case in the future, and it's ok to have (you just
proposed optional in another comment).

[Henrik Grunell, 2016/08/24 12:47:15]

It could be missed when adding a new type. Better to state all possible cases
explicitly and force adding new ones here.

+          "Media.Audio.Render.BufferSizeMismatch.LatencyOther", value);
+  }
+}
+
 // Converts low latency based |output_params| into high latency appropriate
 // output parameters in error situations.
 void AudioOutputResampler::SetupFallbackParams() {
@@ -354,7 +388,9 @@ OnMoreDataConverter::OnMoreDataConverter(const AudioParameters& input_params,
       audio_converter_(input_params, output_params, false),
       error_occurred_(false),
       input_buffer_size_(input_params.frames_per_buffer()),
-      output_buffer_size_(output_params.frames_per_buffer()) {}
+      output_buffer_size_(output_params.frames_per_buffer()) {
+  RecordRebufferingStats(input_params, output_params);
+}
 
 OnMoreDataConverter::~OnMoreDataConverter() {
   // Ensure Stop() has been called so we don't end up with an AudioOutputStream