我正在尝试设置 OpenSL AudioPlayer 以使用我分配的内存来播放 wav 文件。我想这样做,这样我就可以拥有多个共享相同数据并节省内存的 AudioPlayer。
我试图给 openSL 整个文件并告诉它它是带有 format_mime 的 WAVE
SLDataLocator_Address loc_fd = {SL_DATALOCATOR_ADDRESS, data, size};
SLDataFormat_MIME format_mime = { SL_DATAFORMAT_MIME, (SLchar*)"audio/x-wav",SL_CONTAINERTYPE_WAV};
SLDataSource audioSrc = { &loc_fd, &format_mime };
// configure audio sink
SLDataLocator_OutputMix loc_outmix = { SL_DATALOCATOR_OUTPUTMIX,outputMixObject };
SLDataSink audioSnk = { &loc_outmix, 0 };
// create audio player
const SLInterfaceID ids[2] = { SL_IID_SEEK, SL_IID_PLAYBACKRATE };
const SLboolean req[2] = { SL_BOOLEAN_FALSE, SL_BOOLEAN_FALSE };
result = (*engineEngine)->CreateAudioPlayer(engineEngine,&uriPlayerObject[cntSOUND],&audioSrc, &audioSnk, 0, ids, req);
我自己解析了 WAVE 数据并加载了 format_pcm
SLDataFormat_PCM format_pcm;
format_pcm.formatType = SL_DATAFORMAT_PCM;
char* wavParser = isWAVE(data);
if(wavParser == NULL)
{
Log("NOT A WAVE!");
return -1;
}
char* fmtChunk = getChunk("fmt ", data, size);
parsefmtChunk(fmtChunk, &format_pcm);
char* dataChunk = getChunk("data",data, size);
dataChunk += 4;
unsigned int dataSize = *((unsigned int*)dataChunk);
dataChunk += 4;
format_pcm.channelMask = 0;
format_pcm.containerSize = 16;
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
loc_fd.pAddress = dataChunk;
loc_fd.length = dataSize;
parsefmtChunk 函数是
void parsefmtChunk(char* fmtchunk, SLDataFormat_PCM* pcm)
{
char* data = fmtchunk + 8;
unsigned short audioFormat = *((unsigned short*)data);
if(audioFormat != 1)
{
Log("Not PCM!");
Log("Reached Line:%d in File %s", __LINE__, __FILE__);
return;
}
data += 2;
pcm->numChannels = *((unsigned short*)data);
data += 2;
pcm->samplesPerSec = *((unsigned int*)data);
data += 4;
//Byte Rate
data += 4;
//Block Align
data += 2;
//BitsPerSample
pcm->bitsPerSample = *((unsigned short*)data);
(是否应该以某种方式使用字节速率和块对齐来填充 pcm 结构?)
但是每当我创建音频播放器时,我都会得到 SL_RESULT_CONTENT_UNSUPPORTED
这是我从我的 parsefmt 函数 Channels:2 samplesPerSec:44100 bitsPerSample:16 中记录的内容