ALSA相關

音頻數據流向：
| DMA | | I2S/PCM/AC97 |

RAM -------------> I2SControllerFIFO ---------------------------> CODEC ----> SPK/Headset

PCM模塊初始化：

struct snd_soc_platform s3c_soc_platform = {  
           .name         = "s3c-pcm-audio",  
           .pcm_ops      = &s3c_pcm_ops,  //OK
           .pcm_new      = s3c_pcm_new,   //OK
           .pcm_free     = s3c_pcm_free_dma_buffers,  //OK
           .suspend      = s3c_pcm_suspend,  
           .resume       = s3c_pcm_resume,  
    };

調用snd_soc_register_platform()向ALSA core註冊一個snd_soc_platform結構體。成員pcm_new需要調用dma_alloc_writecombine()給DMA分配一塊write-combining的內存空間，並把這塊緩衝區的相關信息保存到substream->dma_buffer中，相當於構造函數。pcm_free則相反。

snd_pcm_ops結構體如下：

struct snd_pcm_ops {  
           int (*open)(struct snd_pcm_substream *substream);  //OK
           int (*close)(struct snd_pcm_substream *substream); //OK 
           int (*ioctl)(struct snd_pcm_substream * substream, unsigned int cmd, void *arg);  
           int (*hw_params)(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params);      //OK
           int (*hw_free)(struct snd_pcm_substream *substream);   //OK
           int (*prepare)(struct snd_pcm_substream *substream);   //OK
           int (*trigger)(struct snd_pcm_substream *substream, int cmd);  //OK
           snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); //PCM中間層通過調用這個函數來獲取緩衝區的位置。 
           int (*copy)(struct snd_pcm_substream *substream, int channel,  
                      snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count);  
           int (*silence)(struct snd_pcm_substream *substream, int channel,   
                         snd_pcm_uframes_t pos, snd_pcm_uframes_t count);  
           struct page *(*page)(struct snd_pcm_substream *substream, unsigned long offset);  
           int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma);  
           int (*ack)(struct snd_pcm_substream *substream);  
    };

open函數爲PCM模塊設定支持的傳輸模式、數據格式、通道數、period等參數，併爲playback/capture stream分配相應的DMA通道。

static int s3c_pcm_open(struct snd_pcm_substream *substream)  
    {  
           struct snd_soc_pcm_runtime *rtd = substream->private_data;  
           struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;  
           struct snd_pcm_runtime *runtime = substream->runtime;  
           struct audio_stream_a *s = runtime->private_data;  
           int ret;  
      
           if (!cpu_dai->active) {  
                  audio_dma_request(&s[0], audio_dma_callback); //爲playback stream分配DMA，audio_dma_callback，這是dma的中斷函數。
                  audio_dma_request(&s[1], audio_dma_callback); //爲capture stream分配DMA  
           }  
             
           //設定runtime硬件參數，硬件參數要根據芯片的數據手冊來定義。
           snd_soc_set_runtime_hwparams(substream, &s3c_pcm_hardware);  
      
           /* Ensure that buffer size is a multiple of period size */  
           ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);        
           return ret;  
    }

static const struct snd_pcm_hardware s3c_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_LE |
SNDRV_PCM_FMTBIT_U8 |
SNDRV_PCM_FMTBIT_S8,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 128*1024,
.period_bytes_min = PAGE_SIZE,
.period_bytes_max = PAGE_SIZE*2,
.periods_min = 2,
.periods_max = 128,
.fifo_size = 32,
};

上層ALSA lib可以通過接口來獲得這些參數的，如snd_pcm_hw_params_get_buffer_size_max()來取得buffer_bytes_max。 hw_params函數爲substream（每打開一個playback或capture，ALSA core均產生相應的一個substream）設定DMA的源（目的）地址，以及DMA緩衝區的大小。

static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,  
                           struct snd_pcm_hw_params *params)  
{  
       struct snd_pcm_runtime *runtime = substream->runtime;  
       int err = 0;  
       /*dma_buffer是DMA緩衝區，它通過4個字段定義：dma_area、dma_addr、dma_bytes和dma_private。其中dma_area是緩衝區邏輯地址，
       dma_addr是緩衝區的物理地址，dma_bytes是緩衝區的大小，dma_private是ALSA的DMA管理用到的。dma_buffer是在pcm_new()中初始化的。*/
       snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);  
       runtime->dma_bytes = params_buffer_bytes(params);  
       return err;  
}

prepare

當pcm“準備好了”調用該函數。在這裏根據channels、buffer_bytes等來設定DMA傳輸參數，跟具體硬件平臺相關。注：每次調用snd_pcm_prepare()的時候均會調用prepare函數。

trigger

當pcm開始、停止、暫停的時候都會調用trigger函數。

ALSA: Advanced Linux Sound Architecture，它包括內核驅動集合、API庫和工具。用戶層程序直接調用libsound的API庫，不需要打開設備等操作，因此編程者不需要了解底層細節。在嵌入式中，音頻數據傳輸一般用I2S接口，控制一般用I2c或SPI接口。如下僅以嵌入式聲卡爲例，其驅動代碼一般放在sound/soc下面。

struct snd_soc_dai結構體：

/*     Digital Audio Interface runtime data.  
       Holds runtime data for a DAI. */  
    struct snd_soc_dai {  
        /* DAI description */  
        char *name;  //模塊聲卡名稱
        unsigned int id;  
        int ac97_control;  
      
        struct device *dev;  
        void *ac97_pdata;   /* platform_data for the ac97 codec */  
      
        /* DAI callbacks */  
        int (*probe)(struct platform_device *pdev, struct snd_soc_dai *dai);  
        void (*remove)(struct platform_device *pdev, struct snd_soc_dai *dai);  
        int (*suspend)(struct snd_soc_dai *dai);  
        int (*resume)(struct snd_soc_dai *dai);  
      
        /* ops */  
        struct snd_soc_dai_ops *ops; //聲卡操作函數集合指針，實現的有hw_params（硬件參數設定）、digital_mute（靜音操作）、set_fmt（格式配置）等，這些函數的實現均與硬件相關，根據硬件的數據手冊來實現。    
        
        /* DAI capabilities */  
        struct snd_soc_pcm_stream capture;  //錄音參數設定
        struct snd_soc_pcm_stream playback; //播放參數設定，均包含channel數目、PCM_RATE和PCM_FMTBIT等信息。
        unsigned int symmetric_rates:1;     //
      
        /* DAI runtime info */  
        struct snd_pcm_runtime *runtime;  
        struct snd_soc_codec *codec;  
        unsigned int active;  
        unsigned char pop_wait:1;  
        void *dma_data;  
      
        /* DAI private data */  
        void *private_data;  
      
        /* parent platform */  
        struct snd_soc_platform *platform;  
      
        struct list_head list;  
    };

結構體定義範例：

struct snd_soc_dai uda134x_dai = {  
        .name = "UDA134X",  
        /* playback capabilities */  
        .playback = {  
            .stream_name = "Playback",  
            .channels_min = 1,  
            .channels_max = 2,  
            .rates = UDA134X_RATES,  
            .formats = UDA134X_FORMATS,  
        },  
        /* capture capabilities */  
        .capture = {  
            .stream_name = "Capture",  
            .channels_min = 1,  
            .channels_max = 2,  
            .rates = UDA134X_RATES,  
            .formats = UDA134X_FORMATS,  
        },  
        /* pcm operations */  
        .ops = &uda134x_dai_ops,  
    };

/* codec device */  
    struct snd_soc_codec_device {  
        int (*probe)(struct platform_device *pdev);  
        int (*remove)(struct platform_device *pdev);  
        int (*suspend)(struct platform_device *pdev, pm_message_t state);  
        int (*resume)(struct platform_device *pdev);  
    };  
Probe指聲卡的探測與初始化，remove指聲卡的卸載，suspend指聲卡的休眠，resume指聲卡從休眠狀態下恢復。詳細介紹probe函數。

static int uda134x_soc_probe(struct platform_device *pdev)
{
//獲得snd_soc_device結構體
//在聲卡的初始化過程中，其實首先是調用sound/soc/<SOC>下的相關驅動的probe函數，在probe有platform_set_drvdata()的操作，
//這裏有個將指針類型的轉換：(struct snd_soc_device *s) ==> (struct platform_device *)。
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
…
//爲codec分配內存
socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (socdev->card->codec == NULL)
return ret;
codec = socdev->card->codec;
…
//初始化codec
codec->name = "uda134x";
codec->owner = THIS_MODULE;
codec->dai = &uda134x_dai; //指向上面定義好的dai
codec->num_dai = 1;
codec->read = uda134x_read_reg_cache; //控制接口—讀
codec->write = uda134x_write; //控制接口—寫
…
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
…
/* register pcms */
/*創建一個PCM實例以便播放數據流。函數裏重要的是如下兩句：
1. ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card); //create and initialize a soundcard structure
2. ret = soc_new_pcm(socdev, &card->dai_link[i], i);//創建播放流/錄音流的子流，將所有播放流/錄音流的子流操作函數設爲soc_pcm_ops。
*/
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
…
/*將操作集合掛到card->control鏈表上來，這個集合實現了音頻播放時各個參數的設置，主要有.info、.get和.set。
如playback volume control：SOC_DOUBLE_R_TLV("Playback Volume", SNDCARD_REG_L_GAIN, SNDCARD_REG_R_GAIN, 0, 192, 0, digital_tlv)，
其中SNDCARD_REG_L_GAIN和SNDCARD_REG_R_GAIN分別是左右聲道音量增益寄存器偏移。最終要調用的函數都是在soc-core.c裏面的，
這裏只是提供一些跟硬件相關的參數，大爲增加了代碼的複用性。*/
ret = snd_soc_add_controls(codec, uda134x_snd_controls, ARRAY_SIZE(uda134x_snd_controls));
…
/* register card */
ret = snd_soc_init_card(socdev);
}

/* SoC Device - the audio subsystem */  
    struct snd_soc_device {  
        struct device *dev;  
        struct snd_soc_card *card;  
        struct snd_soc_codec_device *codec_dev;  
        void *codec_data;  
    };  
這個結構體用於向內核註冊一個device。初始化一般如下：

static struct snd_soc_device SOC_SNDCARD_snd_devdata = {
.card = &snd_soc_s3c24xx_uda134x,
.codec_dev = &soc_codec_dev_uda134x,//就是CODEC定義的snd_soc_codec_device結構體
.codec_data = &s3c24xx_uda134x, //私有數據，一般存放SNDCARD控制接口信息，如I2C從設備地址等
};

對於module_init，其實用platform_driver_register註冊一個platform_driver結構體的方式也好，還是直接寫一個init也好，都問題不大。前者更貼近Linux的驅動模型。Probe的一般過程如下：

static int s3c24xx_uda134x_probe(struct platform_device *pdev)  
    {              
        s3c24xx_uda134x_snd_devdata.codec_dev = &soc_codec_dev_uda134x;  
        s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);  
        platform_set_drvdata(s3c24xx_uda134x_snd_device, & s3c24xx_uda134x_snd_devdata);  
        s3c24xx_uda134x_snd_devdata.dev = & s3c24xx_uda134x_snd_device->dev;  
        platform_device_add(s3c24xx_uda134x_snd_device); //codec中導出的結構體在這裏註冊 
    }

/* SoC card */  
    struct snd_soc_card {  
        char *name;  
        struct device *dev;  
      
        struct list_head list;  
      
        int instantiated;  
      
        int (*probe)(struct platform_device *pdev);  
        int (*remove)(struct platform_device *pdev);  
      
        /* the pre and post PM functions are used to do any PM work before and 
         * after the codec and DAI's do any PM work. */  
        int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);  
        int (*suspend_post)(struct platform_device *pdev, pm_message_t state);  
        int (*resume_pre)(struct platform_device *pdev);  
        int (*resume_post)(struct platform_device *pdev);  
      
        /* callbacks */  
        int (*set_bias_level)(struct snd_soc_card *,  
                      enum snd_soc_bias_level level);  
      
        /* CPU <--> Codec DAI links  */  
        struct snd_soc_dai_link *dai_link;  
        int num_links;  
      
        struct snd_soc_device *socdev;  
      
        struct snd_soc_codec *codec;  
      
        struct snd_soc_platform *platform;  
        struct delayed_work delayed_work;  
        struct work_struct deferred_resume_work;  
    };  
定義這個結構體是讓snd_soc_register_card()註冊一個card的。初始化範例：

static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
.name = "S3C24XX_UDA134X",
.platform = &s3c24xx_soc_platform,
.dai_link = &s3c24xx_uda134x_dai_link,
.num_links = 1,
};

/* SoC machine DAI configuration, glues a codec and cpu DAI together */  
    struct snd_soc_dai_link  {  
        char *name;         /* Codec name */  
        char *stream_name;      /* Stream name */  
      
        /* DAI */  
        struct snd_soc_dai *codec_dai;  
        struct snd_soc_dai *cpu_dai;  
      
        /* machine stream operations */  
        struct snd_soc_ops *ops;  
      
        /* codec/machine specific init - e.g. add machine controls */  
        int (*init)(struct snd_soc_codec *codec);  
      
        /* Symmetry requirements */  
        unsigned int symmetric_rates:1;  
      
        /* Symmetry data - only valid if symmetry is being enforced */  
        unsigned int rate;  
      
        /* DAI pcm */  
        struct snd_pcm *pcm;  
    };  
因爲一個平臺可以運行多個音頻設備，snd_soc_dai_link的作用也在這，將CODEC定義的snd_soc_dai掛到一個鏈表上。
name指定codec名稱；.codec_dai指向CODEC定義的snd_soc_dai結構體；.cpu_dai指向I2S定義的snd_soc_dai結構體；.ops接下來分析。

/* SoC audio ops */
struct snd_soc_ops {
int (*startup)(struct snd_pcm_substream *);
void (*shutdown)(struct snd_pcm_substream *);
int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
int (*hw_free)(struct snd_pcm_substream *);
int (*prepare)(struct snd_pcm_substream *);
int (*trigger)(struct snd_pcm_substream *, int);
};

底層硬件操作—

CODEC：控制接口及芯片基本初始化

PCM：pcm dma操作

I2S：i2s配置操作

之後i2s和pcm其實都跟codec差不多了，只需要理解alsa-core、<soc>、<codec、pcm、i2s>三層的關係。其中codec、pcm、i2s可以看做同層的，分別對於音頻設備的control、dma、i2s接口；<codec、pcm、i2s>會分別export相關結構體給<soc>層，<soc>層將音頻設備三部分與CPU Spec聯結起來，其probe順序是<SOC>.probe-><codec, pcm, i2s>.probe；另外<codec、pcm、i2s>在各自的module_init中將自身註冊到alsa-core中。

模塊測試函數基於linux2.6

源碼控制工具

ALSA各文件作用

ubuntu10.10 later install jdk6

ALSA相關

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