ppsspp/android/jni/native-audio-so.cpp
2019-09-17 23:41:47 +02:00

189 lines
6.1 KiB
C++

// Minimal audio streaming using OpenSL.
//
// Loosely based on the Android NDK sample code.
#include <assert.h>
#include <string.h>
#include <unistd.h>
// for native audio
#include <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
#include "../base/logging.h"
#include "native-audio-so.h"
AudioContext::AudioContext(AndroidAudioCallback cb, int _FramesPerBuffer, int _SampleRate)
: audioCallback(cb), framesPerBuffer(_FramesPerBuffer), sampleRate(_SampleRate) {
if (framesPerBuffer == 0)
framesPerBuffer = 256;
if (framesPerBuffer < 32)
framesPerBuffer = 32;
if (framesPerBuffer > 4096)
framesPerBuffer = 4096;
sampleRate = _SampleRate;
}
// This callback handler is called every time a buffer finishes playing.
// The documentation available is very unclear about how to best manage buffers.
// I've chosen to this approach: Instantly enqueue a buffer that was rendered to the last time,
// and then render the next. Hopefully it's okay to spend time in this callback after having enqueued.
void OpenSLContext::bqPlayerCallbackWrap(SLAndroidSimpleBufferQueueItf bq, void *context) {
OpenSLContext *ctx = (OpenSLContext *)context;
ctx->BqPlayerCallback(bq);
}
void OpenSLContext::BqPlayerCallback(SLAndroidSimpleBufferQueueItf bq) {
if (bq != bqPlayerBufferQueue) {
ELOG("Wrong bq!");
return;
}
int renderedFrames = audioCallback(buffer[curBuffer], framesPerBuffer);
int sizeInBytes = framesPerBuffer * 2 * sizeof(short);
int byteCount = (framesPerBuffer - renderedFrames) * 4;
if (byteCount > 0) {
memset(buffer[curBuffer] + renderedFrames * 2, 0, byteCount);
}
SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[curBuffer], sizeInBytes);
// Comment from sample code:
// the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,
// which for this code example would indicate a programming error
if (result != SL_RESULT_SUCCESS) {
ELOG("OpenSL ES: Failed to enqueue! %i %i", renderedFrames, sizeInBytes);
}
curBuffer += 1; // Switch buffer
if (curBuffer == NUM_BUFFERS)
curBuffer = 0;
}
// create the engine and output mix objects
OpenSLContext::OpenSLContext(AndroidAudioCallback cb, int _FramesPerBuffer, int _SampleRate)
: AudioContext(cb, _FramesPerBuffer, _SampleRate) {}
bool OpenSLContext::Init() {
SLresult result;
// create engine
result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
if (result != SL_RESULT_SUCCESS) {
ELOG("OpenSL ES: Failed to create the engine: %d", (int)result);
return false;
}
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
assert(SL_RESULT_SUCCESS == result);
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 0, 0, 0);
assert(SL_RESULT_SUCCESS == result);
result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
// The constants, such as SL_SAMPLINGRATE_44_1, are just 44100000.
SLuint32 sr = (SLuint32)sampleRate * 1000;
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, NUM_BUFFERS};
SLDataFormat_PCM format_pcm = {
SL_DATAFORMAT_PCM,
2,
sr,
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};
// configure audio sink
SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
SLDataSink audioSnk = {&loc_outmix, NULL};
// create audio player
const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
if (result != SL_RESULT_SUCCESS) {
ELOG("OpenSL ES: CreateAudioPlayer failed: %d", (int)result);
// Should really tear everything down here. Sigh.
return false;
}
result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
&bqPlayerBufferQueue);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, &bqPlayerCallbackWrap, this);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
assert(SL_RESULT_SUCCESS == result);
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
assert(SL_RESULT_SUCCESS == result);
// Allocate and enqueue N empty buffers.
for (int i = 0; i < NUM_BUFFERS; i++) {
buffer[i] = new short[framesPerBuffer * 2]{};
}
int sizeInBytes = framesPerBuffer * 2 * sizeof(short);
for (int i = 0; i < NUM_BUFFERS; i++) {
result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer[i], sizeInBytes);
if (SL_RESULT_SUCCESS != result) {
return false;
}
}
curBuffer = 0;
return true;
}
// shut down the native audio system
OpenSLContext::~OpenSLContext() {
if (bqPlayerPlay) {
ILOG("OpenSLWrap_Shutdown - stopping playback");
SLresult result;
result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
if (SL_RESULT_SUCCESS != result) {
ELOG("SetPlayState failed");
}
}
ILOG("OpenSLWrap_Shutdown - deleting player object");
if (bqPlayerObject) {
(*bqPlayerObject)->Destroy(bqPlayerObject);
bqPlayerObject = nullptr;
bqPlayerPlay = nullptr;
bqPlayerBufferQueue = nullptr;
bqPlayerVolume = nullptr;
}
ILOG("OpenSLWrap_Shutdown - deleting mix object");
if (outputMixObject) {
(*outputMixObject)->Destroy(outputMixObject);
outputMixObject = nullptr;
}
ILOG("OpenSLWrap_Shutdown - deleting engine object");
if (engineObject) {
(*engineObject)->Destroy(engineObject);
engineObject = nullptr;
engineEngine = nullptr;
}
for (int i = 0; i < NUM_BUFFERS; i++) {
delete[] buffer[i];
buffer[i] = nullptr;
}
ILOG("OpenSLWrap_Shutdown - finished");
}