SDL_OpenAudioDevice:从实时处理的源缓冲区连续播放

问题描述 投票:0回答:1

我正在将仿真器移植到SDL。有一种方法在每个帧处调用,该方法传递带有下一个帧的新音频样本的缓冲区。

我用SDL_OpenAudioDevice打开了一个设备,并且在每帧SDL回调方法都会从音频缓冲区中复制样本。

它可以工作,但是声音不是很完美,有些抽打,有些金属噪音等等。

声音是16位带符号的。

编辑:好的,我找到了解决方案!

使用开头帖子的代码,我正在实时播放当前帧的下一帧样本。错了!

因此,我实现了一个循环缓冲区,在其中放置下一帧的样本,基础代码在每个(当前)帧传递给我。

在该缓冲区中,有2个指针,一个指针用于读取点,另一个指针用于写入点。当SDL的音频流上没有更多数据可播放时,它将调用回调函数。因此,当调用回调函数时,我将从循环缓冲区的读取点播放音频样本,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,我将它们写入写点的循环缓冲区中,然后更新写指针。

读取和写入指针移动每帧要播放的样本数量。

代码已更新,当samplesPerFrame不是int但可以工作时,需要进行一些调整;-)

循环缓冲区结构:

typedef struct circularBufferStruct
{
    short *buffer;
    int cells;
    short *readPoint;
    short *writePoint;
} circularBuffer;

此方法在初始化时被调用:

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;
}

这是来自want.callback的SDL方法回调:

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;
}

在每一帧都调用此方法(在第一遍之后,我们只需要采样数量):

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;
}

我希望这个问题/答案将来对其他人有用,因为我在网上几乎找不到SDL Audio的任何内容

c audio sdl-2
1个回答
0
投票

好,我找到了解决方案!

使用开头帖子的代码,我正在实时播放当前帧的下一帧样本。错了!

因此,我实现了一个循环缓冲区,在其中放置下一帧的样本,基础代码在每个(当前)帧传递给我。我从那个缓冲区中的不同位置读写,请参见开头]

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