SDL_OpenAudioDevice:从实时处理的源缓冲区连续播放
我正在写一个模拟器到SDL的移植。有一种方法,在每一帧调用,为下一帧传递带有新音频样本的缓冲区 我用SDL_OpenAudioDevice打开了一个设备,在每一帧SDL回调方法都会从音频缓冲区复制样本 它可以工作,但声音并不完美,有些抽搐,一些金属噪音等等 声音是16位有符号的 编辑:好的,我找到了一个解决方案强> 用开场白的代码,我在当前帧实时播放下一帧的样本。这是错误的 所以,我实现了一个循环缓冲区,在该缓冲区中,我放置了底层代码在每个(当前)帧传递给我的下一帧的样本 在该缓冲区中有两个指针,一个用于读取点,另一个用于写入点。SDL在其音频流上没有更多数据可播放时调用回调函数;所以,当调用回调函数时,我在循环缓冲区上播放读取点的音频样本,然后更新读取指针 当底层代码为我提供下一帧的音频样本数据时,我会在写入点将它们写入循环缓冲区,然后更新写入指针 读和写指针根据每帧要播放的样本量进行移位 代码已更新,当SamplePerFrame不是int但可以工作时需要进行一些调整;-) 循环缓冲结构:SDL_OpenAudioDevice:从实时处理的源缓冲区连续播放,c,audio,sdl-2,C,Audio,Sdl 2,我正在写一个模拟器到SDL的移植。有一种方法,在每一帧调用,为下一帧传递带有新音频样本的缓冲区 我用SDL_OpenAudioDevice打开了一个设备,在每一帧SDL回调方法都会从音频缓冲区复制样本 它可以工作,但声音并不完美,有些抽搐,一些金属噪音等等 声音是16位有符号的 编辑:好的,我找到了一个解决方案 用开场白的代码,我在当前帧实时播放下一帧的样本。这是错误的 所以,我实现了一个循环缓冲区,在该缓冲区中,我放置了底层代码在每个(当前)帧传递给我的下一帧的样本 在该缓冲区中有两个指针,一
typedef struct circularBufferStruct
{
short *buffer;
int cells;
short *readPoint;
short *writePoint;
} circularBuffer;
在初始化时调用此方法:
int initialize_audio(int stereo)
{
if (stereo)
channel = 2;
else
channel = 1;
// Check if sound is disabled
if (sampleRate != 0)
{
// Initialize SDL Audio
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
SDL_Log("SDL fails to initialize audio subsystem!\n%s", SDL_GetError());
return 1;
}
// Number of samples per frame
samplesPerFrame = (double)sampleRate / (double)framesPerSecond * channel;
audioSamplesSize = samplesPerFrame * bytesPerSample; // Bytes
audioBufferSize = audioSamplesSize * 10; // Bytes
// Set and clear circular buffer
audioBuffer.buffer = malloc(audioBufferSize); // Bytes, must be a multiple of audioSamplesSize
memset(audioBuffer.buffer, 0, audioBufferSize);
audioBuffer.cells = (audioBufferSize) / sizeof(short); // Cells, not Bytes!
audioBuffer.readPoint = audioBuffer.buffer;
audioBuffer.writePoint = audioBuffer.readPoint + (short)samplesPerFrame;
}
else
samplesPerFrame = 0;
// First frame
return samplesPerFrame;
}
在每个帧调用此方法(第一次通过后,我们只需要样本量):
我希望这个问题/答案将来会对其他人有用,因为我在网上几乎找不到SDL Audio的任何内容强> 用开场白的代码,我在当前帧实时播放下一帧的样本。这是错误的 所以,我实现了一个循环缓冲区,在该缓冲区中,我放置了底层代码在每个(当前)帧传递给我的下一帧的样本。从那个缓冲区,我在不同的位置读写,见开篇文章
void audioCallback(void *userdata, uint8_t *stream, int len)
{
SDL_memset(stream, 0, len);
if (audioSamplesSize == 0)
return;
if (len > audioSamplesSize)
{
len = audioSamplesSize;
}
SDL_MixAudioFormat(stream, (const Uint8 *)audioBuffer.readPoint, AUDIO_S16SYS, len, SDL_MIX_MAXVOLUME);
audioBuffer.readPoint += (short)samplesPerFrame;
if (audioBuffer.readPoint >= audioBuffer.buffer + audioBuffer.cells)
audioBuffer.readPoint = audioBuffer.readPoint - audioBuffer.cells;
}
int update_audio(short *buffer)
{
// Check if sound is disabled
if (sampleRate != 0)
{
memcpy(audioBuffer.writePoint, buffer, audioSamplesSize); // Bytes
audioBuffer.writePoint += (short)samplesPerFrame; // Cells
if (audioBuffer.writePoint >= audioBuffer.buffer + audioBuffer.cells)
audioBuffer.writePoint = audioBuffer.writePoint - audioBuffer.cells;
if (firstTime)
{
// Set required audio specs
want.freq = sampleRate;
want.format = AUDIO_S16SYS;
want.channels = channel;
want.samples = samplesPerFrame / channel; // total samples divided by channel count
want.padding = 0;
want.callback = audioCallback;
want.userdata = NULL;
device = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(0, 0), 0, &want, &have, 0);
SDL_PauseAudioDevice(device, 0);
firstTime = 0;
}
}
else
samplesPerFrame = 0;
// Next frame
return samplesPerFrame;
}