Android: Ask system for optimal audio buffer size and sample rate

This can reduce audio latency according to
https://developer.android.com/ndk/guides/audio/opensl/opensl-prog-notes#perform.

Previously we were using the hardcoded values of 48000 Hz and 256 frames
per buffer. The sample rate we use with this change is 48000 Hz on all
devices I'm aware of, but the buffer size does vary across devices.

Terminology note: The old code used the term "sample" to refer to what
Android refers to as a "frame". "Frame" is a clearer term to use for
this, so I've changed OpenSLESStream's terminology. One frame consists
of one sample per channel.

Source/Android/app/src/main/java/org/dolphinemu/dolphinemu/utils/AudioUtils.kt

@@ -0,0 +1,28 @@
+package org.dolphinemu.dolphinemu.utils
+
+import android.content.Context
+import android.media.AudioManager
+import androidx.annotation.Keep
+import org.dolphinemu.dolphinemu.DolphinApplication
+
+object AudioUtils {
+    @JvmStatic @Keep
+    fun getSampleRate(): Int =
+        getAudioServiceProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE, 48000)
+
+    @JvmStatic @Keep
+    fun getFramesPerBuffer(): Int =
+        getAudioServiceProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER, 256)
+
+    private fun getAudioServiceProperty(property: String, fallback: Int): Int {
+        return try {
+            val context = DolphinApplication.getAppContext()
+            val am = context.getSystemService(Context.AUDIO_SERVICE) as AudioManager
+            Integer.parseUnsignedInt(am.getProperty(property))
+        } catch (e: NullPointerException) {
+            fallback
+        } catch (e: NumberFormatException) {
+            fallback
+        }
+    }
+}

Source/Android/jni/AndroidCommon/IDCache.cpp

@@ -110,6 +110,10 @@ static jclass s_core_device_control_class;
 static jfieldID s_core_device_control_pointer;
 static jmethodID s_core_device_control_constructor;
 
+static jclass s_audio_utils_class;
+static jmethodID s_audio_utils_get_sample_rate;
+static jmethodID s_audio_utils_get_frames_per_buffer;
+
 static jmethodID s_runnable_run;
 
 namespace IDCache
@@ -517,6 +521,21 @@ jmethodID GetRunnableRun()
   return s_runnable_run;
 }
 
+jclass GetAudioUtilsClass()
+{
+  return s_audio_utils_class;
+}
+
+jmethodID GetAudioUtilsGetSampleRate()
+{
+  return s_audio_utils_get_sample_rate;
+}
+
+jmethodID GetAudioUtilsGetFramesPerBuffer()
+{
+  return s_audio_utils_get_frames_per_buffer;
+}
+
 }  // namespace IDCache
 
 extern "C" {
@@ -724,6 +743,13 @@ JNIEXPORT jint JNI_OnLoad(JavaVM* vm, void* reserved)
                        "(Lorg/dolphinemu/dolphinemu/features/input/model/CoreDevice;J)V");
   env->DeleteLocalRef(core_device_control_class);
 
+  const jclass audio_utils_class = env->FindClass("org/dolphinemu/dolphinemu/utils/AudioUtils");
+  s_audio_utils_class = reinterpret_cast<jclass>(env->NewGlobalRef(audio_utils_class));
+  s_audio_utils_get_sample_rate = env->GetStaticMethodID(audio_utils_class, "getSampleRate", "()I");
+  s_audio_utils_get_frames_per_buffer =
+      env->GetStaticMethodID(audio_utils_class, "getFramesPerBuffer", "()I");
+  env->DeleteLocalRef(audio_utils_class);
+
   const jclass runnable_class = env->FindClass("java/lang/Runnable");
   s_runnable_run = env->GetMethodID(runnable_class, "run", "()V");
   env->DeleteLocalRef(runnable_class);
@@ -761,5 +787,6 @@ JNIEXPORT void JNI_OnUnload(JavaVM* vm, void* reserved)
   env->DeleteGlobalRef(s_numeric_setting_class);
   env->DeleteGlobalRef(s_core_device_class);
   env->DeleteGlobalRef(s_core_device_control_class);
+  env->DeleteGlobalRef(s_audio_utils_class);
 }
 }

Source/Android/jni/AndroidCommon/IDCache.h

@@ -111,4 +111,8 @@ jmethodID GetCoreDeviceControlConstructor();
 
 jmethodID GetRunnableRun();
 
+jclass GetAudioUtilsClass();
+jmethodID GetAudioUtilsGetSampleRate();
+jmethodID GetAudioUtilsGetFramesPerBuffer();
+
 }  // namespace IDCache

Source/Core/AudioCommon/OpenSLESStream.cpp

@@ -8,11 +8,13 @@
 
 #include <SLES/OpenSLES.h>
 #include <SLES/OpenSLES_Android.h>
+#include <jni.h>
 
 #include "Common/Assert.h"
 #include "Common/CommonTypes.h"
 #include "Common/Logging/Log.h"
 #include "Core/ConfigManager.h"
+#include "jni/AndroidCommon/IDCache.h"
 
 void OpenSLESStream::BQPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void* context)
 {
@@ -24,8 +26,8 @@ void OpenSLESStream::PushSamples(SLAndroidSimpleBufferQueueItf bq)
   ASSERT(bq == m_bq_player_buffer_queue);
 
   // Render to the fresh buffer
-  m_mixer->Mix(reinterpret_cast<short*>(m_buffer[m_current_buffer]), BUFFER_SIZE_IN_SAMPLES);
-  SLresult result = (*bq)->Enqueue(bq, m_buffer[m_current_buffer], sizeof(m_buffer[0]));
+  m_mixer->Mix(m_buffer[m_current_buffer].data(), m_frames_per_buffer);
+  SLresult result = (*bq)->Enqueue(bq, m_buffer[m_current_buffer].data(), m_bytes_per_buffer);
   m_current_buffer ^= 1;  // Switch buffer
 
   // Comment from sample code:
@@ -36,6 +38,23 @@ void OpenSLESStream::PushSamples(SLAndroidSimpleBufferQueueItf bq)
 
 bool OpenSLESStream::Init()
 {
+  JNIEnv* env = IDCache::GetEnvForThread();
+  jclass audio_utils = IDCache::GetAudioUtilsClass();
+  const SLuint32 sample_rate =
+      env->CallStaticIntMethod(audio_utils, IDCache::GetAudioUtilsGetSampleRate());
+  m_frames_per_buffer =
+      env->CallStaticIntMethod(audio_utils, IDCache::GetAudioUtilsGetFramesPerBuffer());
+
+  INFO_LOG_FMT(AUDIO, "OpenSLES configuration: {} Hz, {} frames per buffer", sample_rate,
+               m_frames_per_buffer);
+
+  constexpr SLuint32 channels = 2;
+  const SLuint32 samples_per_buffer = m_frames_per_buffer * channels;
+  m_bytes_per_buffer = m_frames_per_buffer * channels * sizeof(m_buffer[0][0]);
+
+  for (std::vector<short>& buffer : m_buffer)
+    buffer.resize(samples_per_buffer);
+
   SLresult result;
   // create engine
   result = slCreateEngine(&m_engine_object, 0, nullptr, 0, nullptr, nullptr);
@@ -50,13 +69,11 @@ bool OpenSLESStream::Init()
   ASSERT(SL_RESULT_SUCCESS == result);
 
   SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
-  SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM,
-                                 2,
-                                 m_mixer->GetSampleRate() * 1000,
-                                 SL_PCMSAMPLEFORMAT_FIXED_16,
-                                 SL_PCMSAMPLEFORMAT_FIXED_16,
-                                 SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
-                                 SL_BYTEORDER_LITTLEENDIAN};
+  SLDataFormat_PCM format_pcm = {
+      SL_DATAFORMAT_PCM,           channels,
+      sample_rate * 1000,          SL_PCMSAMPLEFORMAT_FIXED_16,
+      SL_PCMSAMPLEFORMAT_FIXED_16, SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
+      SL_BYTEORDER_LITTLEENDIAN};
 
   SLDataSource audioSrc = {&loc_bufq, &format_pcm};
 
@@ -92,7 +109,7 @@ bool OpenSLESStream::Init()
   m_current_buffer ^= 1;
 
   result = (*m_bq_player_buffer_queue)
-               ->Enqueue(m_bq_player_buffer_queue, m_buffer[0], sizeof(m_buffer[0]));
+               ->Enqueue(m_bq_player_buffer_queue, m_buffer[0].data(), m_bytes_per_buffer);
   if (SL_RESULT_SUCCESS != result)
     return false;
 

Source/Core/AudioCommon/OpenSLESStream.h

@@ -4,6 +4,9 @@
 #pragma once
 
 #ifdef HAVE_OPENSL_ES
+#include <array>
+#include <vector>
+
 #include <SLES/OpenSLES.h>
 #include <SLES/OpenSLES_Android.h>
 #endif  // HAVE_OPENSL_ES
@@ -35,11 +38,11 @@ private:
   SLAndroidSimpleBufferQueueItf m_bq_player_buffer_queue;
   SLVolumeItf m_bq_player_volume;
 
-  static constexpr int BUFFER_SIZE = 512;
-  static constexpr int BUFFER_SIZE_IN_SAMPLES = BUFFER_SIZE / 2;
+  SLuint32 m_frames_per_buffer;
+  SLuint32 m_bytes_per_buffer;
 
   // Double buffering.
-  short m_buffer[2][BUFFER_SIZE];
+  std::array<std::vector<short>, 2> m_buffer;
   int m_current_buffer = 0;
 #endif  // HAVE_OPENSL_ES
 };