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

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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.
This commit is contained in:
JosJuice 2024-06-06 15:38:41 +02:00
parent 34e8fb068f
commit f99d3dbd5c
5 changed files with 92 additions and 13 deletions
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37
Source/Core/AudioCommon/OpenSLESStream.cpp
View file

@ -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;