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Unified Diff: media/audio/audio_output_resampler.cc

Issue 2268253002: UMA stats for browser/renderer audio rendering buffer size mismatch. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: build fix Created 4 years, 4 months ago
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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..2b70abf5aac63ecf1a03fe5ee990a5ff01127242 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();
Ilya Sherman 2016/08/24 22:20:23 nit: An int is generally going to be smaller or eq
o1ka 2016/08/25 11:48:34 It's extending lifetime of a temporary rather than
+ DCHECK(input_buffer_size);
+ DCHECK(output_buffer_size);
+ int value = (output_buffer_size - input_buffer_size) * 1000 /
+ std::min(input_buffer_size, output_buffer_size);
Ilya Sherman 2016/08/24 22:20:23 What is the possible range for these values? It's
o1ka 2016/08/25 11:48:34 For both buffer sizes the range is between 128 and
Ilya Sherman 2016/08/26 00:35:34 I think you are trying to pack too much into a sin
+
+ 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:
Ilya Sherman 2016/08/24 22:20:23 Please explicitly list all possible cases. AFAICT
o1ka 2016/08/25 11:48:34 Agree. Done
+ UMA_HISTOGRAM_SPARSE_SLOWLY(
+ "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
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