Linux 获取音频ALSA API编程

ALSA Programming HOWTO

根据ALSA写一简单的PCM应用程序,我们首先需要为PCM设备打开一个句柄(Handle),然后指定PCM流的方向是播放或者是捕获(playback还是capture),我们也可以配置一些我们想要的参数,比如,buffer size, sample rate, pcm数据格式等.因此我们就有了一个大体的框架了,简单高效,如下:

/* Handle for the PCM device */ 声明一个设备句柄.
    snd_pcm_t *pcm_handle;          

    /* Playback stream */ 设置音频流的方向为播放
    snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;

    /* This structure contains information about    */声明一个硬件配置结构,用来配置相应的参数.
    /* the hardware and can be used to specify the  */      
    /* configuration to be used for the PCM stream. */ 
    snd_pcm_hw_params_t *hwparams;

PCM设备在ALSA的接口有"plughw"和"hw",接口,不过我们这里使用默认的接口"default",

char *pcm_name;//定义设备名称

pcm_name = strdup("default");

然后我们分配一个硬件配置结构体

/* Allocate the snd_pcm_hw_params_t structure on the stack. */
    snd_pcm_hw_params_alloca(&hwparams);

现在我们可以打开设备了,

/* Open PCM. The last parameter of this function is the mode. */
    /* If this is set to 0, the standard mode is used. Possible   */
    /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC.       */ 
    /* If SND_PCM_NONBLOCK is used, read / write access to the    */
    /* PCM device will return immediately. If SND_PCM_ASYNC is    */
    /* specified, SIGIO will be emitted whenever a period has     */
    /* been completely processed by the soundcard.                */
    if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0) {
      fprintf(stderr, "Error opening PCM device %s\n", pcm_name);
      return(-1);
    }

接下来的部分才是最重要的,前面基本上都是流程化的东西,接下来要说的是对设备的一些配置参数,搞明白这里我觉得比较重要,要配置的参数有 : 访问类型,采样率, 采样格式, 通道数, period的数量以及period的大小,这些参数在上一节中都有详细的介绍,如还有疑问可以点击:http://blog.csdn.net/liuchen_csdn/article/details/52095813
初始化配置硬件的结构体,alsa已经给出了相应的函数,我们搬用就可以了:

/* Init hwparams with full configuration space */
    if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
      fprintf(stderr, "Can not configure this PCM device.\n");
      return(-1);
    }

下面举个例子:我们假设声卡要配置的格式为: 立体声 16Bit Little Endian data, sampled at 44100HZ,接下来我们按照这个参数来配置我们的声卡:

int rate = 44100; /* Sample rate */
    int exact_rate;   /* Sample rate returned by */
                      /* snd_pcm_hw_params_set_rate_near */ 
    int dir;          /* exact_rate == rate --> dir = 0 */
                      /* exact_rate < rate  --> dir = -1 */
                      /* exact_rate > rate  --> dir = 1 */
    int periods = 2;       /* Number of periods */
    snd_pcm_uframes_t periodsize = 8192; /* Periodsize (bytes) */

上面介绍的配置参数中有访问类型这个参数,这个参数指定了多路数据在buffer中的存储方式,分为两种:INTERLEAVED 和 NONINTERLEAVED,对于前者,它的意思就是在每个周期(period)左右声道的数据交叉存放,而对于后者,它是在每个周期(period)先采集左声道的数据存放在buffer中,接着采样右声道的数据在buffer中,例子如下:

 /* Set access type. This can be either    */
    /* SND_PCM_ACCESS_RW_INTERLEAVED or       */
    /* SND_PCM_ACCESS_RW_NONINTERLEAVED.      */
    /* There are also access types for MMAPed */
    /* access, but this is beyond the scope   */
    /* of this introduction.                  */
    if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
      fprintf(stderr, "Error setting access.\n");
      return(-1);
    }
  
    /* Set sample format */
    if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) {
      fprintf(stderr, "Error setting format.\n");
      return(-1);
    }

    /* Set sample rate. If the exact rate is not supported */
    /* by the hardware, use nearest possible rate.         */ 
    exact_rate = rate;
    if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0) < 0) {
      fprintf(stderr, "Error setting rate.\n");
      return(-1);
    }
    if (rate != exact_rate) {
      fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n 
                       ==> Using %d Hz instead.\n", rate, exact_rate);
    }

    /* Set number of channels */
    if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) < 0) {
      fprintf(stderr, "Error setting channels.\n");
      return(-1);
    }

    /* Set number of periods. Periods used to be called fragments. */ 
    if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) {
      fprintf(stderr, "Error setting periods.\n");
      return(-1);
    }

接着我们要设置buffer的大小,buffer的大小是由函数的设置决定,有时参数是字节数,有时需要指定的是frames的数量,一帧的数据大小包括的是所有通道的数据大小,比如,立体声16-Bit 数据,它的一帧就是4Bytes.

/* Set buffer size (in frames). The resulting latency is given by */
    /* latency = periodsize * periods / (rate * bytes_per_frame)     */
    if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (periodsize * periods)>>2) < 0) {
      fprintf(stderr, "Error setting buffersize.\n");
      return(-1);
    }

对于 (periodsize * periods)>>2 的解释,periodsize这个是每个period的字节数,而periods指定是这个buffer要能够存放的周期个数, >>2的意思就是除以4,为什么要除以四呢? 这是因为每个frame占用的字节数是4Bytes啊,所以这个(periodsize * periods)>>2 的结果就是要一共要分配period个周期要容纳的frames数量.
如果你的硬件buffer不支持2^n,你可以使用snd_pcm_hw_params_set_buffer_size_near.这个函数,它会分配相邻的buffer空间.

当所有参数都配置完成后我们要让这些参数作用于PCM设备:

* Apply HW parameter settings to */
    /* PCM device and prepare device  */
    if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) {
      fprintf(stderr, "Error setting HW params.\n");
      return(-1);
    }

PCM设备配置之后,我们就可以播放数据了,但是这里还要注意的是对于interleaved write access,我们调用:

/* Write num_frames frames from buffer data to    */ 
    /* the PCM device pointed to by pcm_handle.       */
    /* Returns the number of frames actually written. */
    snd_pcm_sframes_t snd_pcm_writei(pcm_handle, data, num_frames);

对于noninterleaved access,我们调用:

 /* Write num_frames frames from buffer data to    */ 
    /* the PCM device pointed to by pcm_handle.       */ 
    /* Returns the number of frames actually written. */
    snd_pcm_sframes_t snd_pcm_writen(pcm_handle, data, num_frames);

<<<<<<<<<<<<<< 以上部分是对于播放部分而言的,下面这部分是针对音频采集部分而言 >>>>>>>>>>>>>>>>>>>>>>

要想同时进行采集和播放音频数据仅仅使用一个PCM设备句柄是不可能的,所以我们要声明两个句柄,这样才能实现同步.下面仅介绍不同点:
[1] : PCM数据流的方向不一样了

/* Capture stream */
    snd_pcm_stream_t stream_capture = SND_PCM_STREAM_CAPTURE;

[2] : 参数的配置和播放一样,此处省略......
[3] : 捕获函数不一样啊:

/* Read num_frames frames from the PCM device  */         
    /* pointed to by pcm_handle to buffer capdata. */      
    /* Returns the number of frames actually read. */   
    snd_pcm_readn(pcm_capture_handle, capdata, num_frames);

[4] :
As in the case of playback, we have to take care that the application calls the read function before the capture buffer of the soundcard is completely filled. Otherwise there will be a buffer overrun.

int pcmreturn; while ((pcmreturn = snd_pcm_readi(pcm_capture_handle, capdata, periodsize>>2)) < 0) { snd_pcm_prepare(pcm_capture_handle); fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Overrun >>>>>>>>>>>>>>>\n"); }

转自：https://blog.csdn.net/liuchen_csdn/article/details/52097088

Linux 获取音频ALSA API编程

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