SDL_AudioSpec結構體分析

首先看源碼：

typedef struct SDL_AudioSpec {
	int freq;		/**< DSP frequency -- samples per second */
	Uint16 format;		/**< Audio data format */
	Uint8  channels;	/**< Number of channels: 1 mono, 2 stereo */
	Uint8  silence;		/**< Audio buffer silence value (calculated) */
	Uint16 samples;		/**< Audio buffer size in samples (power of 2) */
	Uint16 padding;		/**< Necessary for some compile environments */
	Uint32 size;		/**< Audio buffer size in bytes (calculated) */
	/**
	 *  This function is called when the audio device needs more data.
	 *
	 *  @param[out] stream	A pointer to the audio data buffer
	 *  @param[in]  len	The length of the audio buffer in bytes.
	 *
	 *  Once the callback returns, the buffer will no longer be valid.
	 *  Stereo samples are stored in a LRLRLR ordering.
	 */
	void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
	void  *userdata;
} SDL_AudioSpec;

SDL_AudioSpec結構體是用來描述某些音頻數據的格式。 結構體中的變量都會被SDL_OpenAudio（）函數用到，想要用SDL處理音頻數據，必須搞懂這些基礎的東西。

freq 指定了每秒向音頻設備發送的sample數。經常使用的值爲：11025，22050，44100。值越高質量越好。

format 指定了每一個sample元素的大小和類型。可能取值以下：

AUDIO_U8	unsigned 8-bit samples.
AUDIO_S8	signed 8-bit samples.
AUDIO_U16 or AUDIO_U16LSB	並不是全部硬件都支持(unsigned 16-bit little-endian)
AUDIO_S16 or AUDIO_S16LSB	並不是全部硬件都支持 (signed 16-bit little-endian)
AUDIO_U16MSB	並不是全部硬件都支持 (unsigned 16-bit big-endian)
AUDIO_S16MSB	並不是全部硬件都支持 (signed 16-bit big-endian)
int32 support
AUDIO_S32 or AUDIO_S32LSB	32-bit integer samples
AUDIO_S32MSB	as above, but big-endian byte order
float32 support
AUDIO_F32 or AUDIO_F32LSB	32-bit floating point samples
AUDIO_F32MSB	as above, but big-endian byte order
本地音頻字節序
AUDIO_U16SYS	AUDIO_U16LSB or AUDIO_U16MSB 取決於硬件CPU字節序
AUDIO_S16SYS	AUDIO_S16LSB or AUDIO_S16MSB 取決於硬件CPU字節序
AUDIO_S32SYS	AUDIO_S32LSB or AUDIO_S32MSB 取決於硬件CPU字節序
AUDIO_F32SYS	AUDIO_F32LSB or AUDIO_F32MSB 取決於硬件CPU字節序

 

channels 指定了聲音的通道數：1（單聲道）2（立體聲）。

samples 這個值表示音頻緩存區的大小（以sample計）。一個sample是一段大小爲 format * channels 的音頻數據。

size 這個值表示音頻緩存區的大小（以byte計）。

silence 設置靜音的值。

padding 考慮到兼容性的一個參數。

callback 是獲取音頻數據後的回調函數，能夠做解碼獲取的音頻碼流及輸出到設備的操做。例如：

void audio_callback(void *userdata, Uint8 *stream, int len){

    AVCodecContext *aCodecCtx = (AVCodecContext *) userdata;
    int len1, audio_size;

    static uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];
    static unsigned int audio_buf_size = 0;
    static unsigned int audio_buf_index = 0;

    SDL_memset(stream, 0, len);

    printf("audio_callback len=%d \n", len);

    //向設備發送長度爲len的數據
    while(len > 0){
        //緩衝區中無數據
        if(audio_buf_index >= audio_buf_size){
            //從packet中解碼數據
            audio_size = audio_decode_frame(aCodecCtx, audio_buf, audio_buf_size);
            printf("audio_decode_frame finish  audio_size=%d \n", audio_size);
            if(audio_size < 0){//沒有解碼到數據或者出錯，填充0
                audio_buf_size = 1024;
                memset(audio_buf, 0, audio_buf_size);
            }else{
                audio_buf_size = audio_size;
            }

            audio_buf_index = 0;
        }

        len1 = audio_buf_size - audio_buf_index;
        if(len1 > len)
            len1 = len;

        SDL_MixAudio(stream, audio_buf + audio_buf_index, len, SDL_MIX_MAXVOLUME);

        len -= len1;
        stream += len1;
        audio_buf_index += len1;
    }
}

關於callback的參數：

    

userdata	程序特定的參數，保存在SDL_AudioSpec結構體中
stream	指向緩存區的指針，這個緩存區將被這個回調函數填滿
len	上面緩存區的大小（字節數）