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Add UMA histogram for actual Android buffer size

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Previously a histogram was added to track the requested buffer size,
this CL adds a histogram for the actually used buffer size.

Bug: b/157429867
Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31385}
This commit is contained in:
Ivo Creusen
2020-05-28 13:54:49 +02:00
committed by Commit Bot
parent b940a7d97b
commit f1393e23a2
5 changed files with 59 additions and 8 deletions
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28
sdk/android/src/jni/audio_device/audio_track_jni.cc
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@ -20,6 +20,7 @@
#include "sdk/android/generated_java_audio_device_module_native_jni/WebRtcAudioTrack_jni.h"
#include "sdk/android/src/jni/jni_helpers.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@ -89,12 +90,33 @@ int32_t AudioTrackJni::InitPlayout() {
nullptr);
if (buffer_size_factor == 0)
buffer_size_factor = 1.0;
if (!Java_WebRtcAudioTrack_initPlayout(
env_, j_audio_track_, audio_parameters_.sample_rate(),
static_cast<int>(audio_parameters_.channels()), buffer_size_factor)) {
int requested_buffer_size_bytes = Java_WebRtcAudioTrack_initPlayout(
env_, j_audio_track_, audio_parameters_.sample_rate(),
static_cast<int>(audio_parameters_.channels()), buffer_size_factor);
if (requested_buffer_size_bytes < 0) {
RTC_LOG(LS_ERROR) << "InitPlayout failed";
return -1;
}
// Update UMA histograms for both the requested and actual buffer size.
// To avoid division by zero, we assume the sample rate is 48k if an invalid
// value is found.
const int sample_rate = audio_parameters_.sample_rate() <= 0
? 48000
: audio_parameters_.sample_rate();
// This calculation assumes that audio is mono.
const int requested_buffer_size_ms =
(requested_buffer_size_bytes * 1000) / (2 * sample_rate);
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeRequestedAudioBufferSizeMs",
requested_buffer_size_ms, 0, 1000, 100);
int actual_buffer_size_frames =
Java_WebRtcAudioTrack_getBufferSizeInFrames(env_, j_audio_track_);
if (actual_buffer_size_frames >= 0) {
const int actual_buffer_size_ms =
actual_buffer_size_frames * 1000 / sample_rate;
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeAudioBufferSizeMs",
actual_buffer_size_ms, 0, 1000, 100);
}
initialized_ = true;
return 0;
}