Add UMA histogram for actual Android buffer size

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}

modules/audio_device/android/audio_track_jni.cc

@@ -34,7 +34,9 @@ AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
       set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
       get_stream_max_volume_(
           native_reg->GetMethodId("getStreamMaxVolume", "()I")),
-      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {}
+      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")),
+      get_buffer_size_in_frames_(
+          native_reg->GetMethodId("getBufferSizeInFrames", "()I")) {}
 
 AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
 
@@ -46,15 +48,26 @@ bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
              nullptr);
   if (buffer_size_factor == 0)
     buffer_size_factor = 1.0;
-  int buffer_size_bytes = audio_track_->CallIntMethod(
+  int requested_buffer_size_bytes = audio_track_->CallIntMethod(
       init_playout_, sample_rate, channels, buffer_size_factor);
-  if (buffer_size_bytes != -1) {
+  // Update UMA histograms for both the requested and actual buffer size.
+  if (requested_buffer_size_bytes >= 0) {
     // To avoid division by zero, we assume the sample rate is 48k if an invalid
     // value is found.
     sample_rate = sample_rate <= 0 ? 48000 : sample_rate;
-    const int buffer_size_ms = (buffer_size_bytes * 1000) / (2 * 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",
-                         buffer_size_ms, 0, 1000, 100);
+                         requested_buffer_size_ms, 0, 1000, 100);
+    int actual_buffer_size_frames =
+        audio_track_->CallIntMethod(get_buffer_size_in_frames_);
+    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);
+    }
     return true;
   }
   return false;

modules/audio_device/android/audio_track_jni.h

@@ -62,6 +62,7 @@ class AudioTrackJni {
     jmethodID set_stream_volume_;
     jmethodID get_stream_max_volume_;
     jmethodID get_stream_volume_;
+    jmethodID get_buffer_size_in_frames_;
   };
 
   explicit AudioTrackJni(AudioManager* audio_manager);

modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java

@@ -433,6 +433,13 @@ public class WebRtcAudioTrack {
     }
   }
 
+  private int getBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= 23) {
+      return audioTrack.getBufferSizeInFrames();
+    }
+    return -1;
+  }
+
   private void logBufferCapacityInFrames() {
     if (Build.VERSION.SDK_INT >= 24) {
       Logging.d(TAG,

sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java

@@ -423,6 +423,14 @@ class WebRtcAudioTrack {
     }
   }
 
+  @CalledByNative
+  private int getBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      return audioTrack.getBufferSizeInFrames();
+    }
+    return -1;
+  }
+
   private void logBufferCapacityInFrames() {
     if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
       Logging.d(TAG,

sdk/android/src/jni/audio_device/audio_track_jni.cc

@@ -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;
 }