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 |