基于ALSA的WAV播放和录音程序

这段时间在探索ALSA架构，从ALSA Core到ALSA Lib，再到Android Audio System。在看ALSA Lib时，写了一个比较典型的基于ALSA的播放录音程序。程序包包含四个部分：

WAV Parser是对WAV文件的分析和封装，这里只针对Standard WAV File；

SND Common是Playback 和Record共同操作，如SetParams、ReadPCM和WritePCM等；

Playback和Record就分别是播放录音的主体了。

原理很简单，以Playback为例：从WAV文件读取PCM数据，通过I2S或AC97依次送到Audio Codec。难点在于对snd_pcm_hw_params_t的设置，尤其要确定每次要送到Audio Codec的数据帧大小(peroid_size)，这个稍后解释。

1、从WAV文件的头信息可以分析出：sample_format、channels number、sample_rate、sample_length，这些参数要通过snd_pcm_hw_params_set_XXX()接口设置到snd_pcm_hw_params_t中。

2、接着我们要设置buffer_time 和peroid_time。通过snd_pcm_hw_params_get_buffer_time_max()接口可以获取该Audio Codec可以支持的最大buffer_time，这里我们设置buffer_time = (MAX_BUFFER_TIME > 500000) ? 500000 : MAX_BUFFER_TIME; peroid_time = buffer_time/4。

关于peroid的概念有这样的描述：The “period” is a term that corresponds to a fragment in the OSS world. The period defines the size at which a PCM interrupt is generated. 从底层驱动看来，应该是PCM DMA单次传送数据帧的大小。其实真正关注底层驱动的话，它并不是关心peroid_time，它关心的是peroid_size，这两者有转换关系。具体见struct snd_pcm_hardware结构体。

3、通过snd_pcm_hw_params_get_period_size()取得peroid_size，注意在ALSA中peroid_size是以frame为单位的。The configured buffer and period sizes are stored in “frames” in the runtime. 1 frame = channels * sample_size. 所以要对peroid_size进行转换：chunk_bytes = peroid_size * sample_length / 8。chunk_bytes就是我们单次从WAV读PCM数据的大小。

之后的过程就乏善可陈了。唯一要留意的是snd_pcm_writei()和snd_pcm_readi()的第三个参数size也是以frame为单位，不要忘记frames和bytes的转换。

//File   : wav_parser.h
//Author : Loon <[email protected]>

#ifndef __WAV_PARSER_H
#define __WAV_PARSER_H

typedef unsigned char  uint8_t;
typedef unsigned short uint16_t;
typedef unsigned int   uint32_t;

#if __BYTE_ORDER == __LITTLE_ENDIAN
#define COMPOSE_ID(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24))
#define LE_SHORT(v)           (v)
#define LE_INT(v)               (v)
#define BE_SHORT(v)           bswap_16(v)
#define BE_INT(v)               bswap_32(v)
#elif __BYTE_ORDER == __BIG_ENDIAN
#define COMPOSE_ID(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24))
#define LE_SHORT(v)           bswap_16(v)
#define LE_INT(v)               bswap_32(v)
#define BE_SHORT(v)           (v)
#define BE_INT(v)               (v)
#else
#error "Wrong endian"
#endif

#define WAV_RIFF        COMPOSE_ID('R','I','F','F')
#define WAV_WAVE        COMPOSE_ID('W','A','V','E')
#define WAV_FMT         COMPOSE_ID('f','m','t',' ')
#define WAV_DATA        COMPOSE_ID('d','a','t','a')

/* WAVE fmt block constants from Microsoft mmreg.h header */
#define WAV_FMT_PCM             0x0001
#define WAV_FMT_IEEE_FLOAT      0x0003
#define WAV_FMT_DOLBY_AC3_SPDIF 0x0092
#define WAV_FMT_EXTENSIBLE      0xfffe

/* Used with WAV_FMT_EXTENSIBLE format */
#define WAV_GUID_TAG        "/x00/x00/x00/x00/x10/x00/x80/x00/x00/xAA/x00/x38/x9B/x71"

/* it's in chunks like .voc and AMIGA iff, but my source say there
   are in only in this combination, so I combined them in one header;
   it works on all WAVE-file I have
 */
typedef struct WAVHeader {
    uint32_t magic;     /* 'RIFF' */
    uint32_t length;        /* filelen */
    uint32_t type;      /* 'WAVE' */
} WAVHeader_t;

typedef struct WAVFmt {
    uint32_t magic;  /* 'FMT '*/
    uint32_t fmt_size; /* 16 or 18 */
    uint16_t format;        /* see WAV_FMT_* */
    uint16_t channels;
    uint32_t sample_rate;   /* frequence of sample */
    uint32_t bytes_p_second;
    uint16_t blocks_align;  /* samplesize; 1 or 2 bytes */
    uint16_t sample_length; /* 8, 12 or 16 bit */
} WAVFmt_t;

typedef struct WAVFmtExtensible {
    WAVFmt_t format;
    uint16_t ext_size;
    uint16_t bit_p_spl;
    uint32_t channel_mask;
    uint16_t guid_format;   /* WAV_FMT_* */
    uint8_t guid_tag[14];   /* WAV_GUID_TAG */
} WAVFmtExtensible_t;

typedef struct WAVChunkHeader {
    uint32_t type;      /* 'data' */
    uint32_t length;        /* samplecount */
} WAVChunkHeader_t;

typedef struct WAVContainer {
    WAVHeader_t header;
    WAVFmt_t format;
    WAVChunkHeader_t chunk;
} WAVContainer_t;

int WAV_ReadHeader(int fd, WAVContainer_t *container);

int WAV_WriteHeader(int fd, WAVContainer_t *container);

#endif /* #ifndef __WAV_PARSER_H */

//File   : wav_parser.c
//Author : Loon <[email protected]>

#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>

#include "wav_parser.h"

#define WAV_PRINT_MSG

char *WAV_P_FmtString(uint16_t fmt)
{
    switch (fmt) {
    case WAV_FMT_PCM:
        return "PCM";
        break;
    case WAV_FMT_IEEE_FLOAT:
        return "IEEE FLOAT";
        break;
    case WAV_FMT_DOLBY_AC3_SPDIF:
        return "DOLBY AC3 SPDIF";
        break;
    case WAV_FMT_EXTENSIBLE:
        return "EXTENSIBLE";
        break;
    default:
        break;
    }

    return "NON Support Fmt";
}

void WAV_P_PrintHeader(WAVContainer_t *container)
{
    printf("+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++/n");
    printf("/n");

    printf("File Magic:         [%c%c%c%c]/n",
        (char)(container->header.magic),
        (char)(container->header.magic>>8),
        (char)(container->header.magic>>16),
        (char)(container->header.magic>>24));
    printf("File Length:        [%d]/n", container->header.length);
    printf("File Type:          [%c%c%c%c]/n",
        (char)(container->header.type),
        (char)(container->header.type>>8),
        (char)(container->header.type>>16),
        (char)(container->header.type>>24));

    printf("/n");

    printf("Fmt Magic:          [%c%c%c%c]/n",
        (char)(container->format.magic),
        (char)(container->format.magic>>8),
        (char)(container->format.magic>>16),
        (char)(container->format.magic>>24));
    printf("Fmt Size:           [%d]/n", container->format.fmt_size);
    printf("Fmt Format:         [%s]/n", WAV_P_FmtString(container->format.format));
    printf("Fmt Channels:       [%d]/n", container->format.channels);
    printf("Fmt Sample_rate:    [%d](HZ)/n", container->format.sample_rate);
    printf("Fmt Bytes_p_second: [%d]/n", container->format.bytes_p_second);
    printf("Fmt Blocks_align:   [%d]/n", container->format.blocks_align);
    printf("Fmt Sample_length:  [%d]/n", container->format.sample_length);

    printf("/n");

    printf("Chunk Type:         [%c%c%c%c]/n",
        (char)(container->chunk.type),
        (char)(container->chunk.type>>8),
        (char)(container->chunk.type>>16),
        (char)(container->chunk.type>>24));
    printf("Chunk Length:       [%d]/n", container->chunk.length);

    printf("/n");
    printf("+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++/n");
}

int WAV_P_CheckValid(WAVContainer_t *container)
{
    if (container->header.magic != WAV_RIFF ||
        container->header.type != WAV_WAVE ||
        container->format.magic != WAV_FMT ||
        container->format.fmt_size != LE_INT(16) ||
        (container->format.channels != LE_SHORT(1) && container->format.channels != LE_SHORT(2)) ||
        container->chunk.type != WAV_DATA) {

        fprintf(stderr, "non standard wav file./n");
        return -1;
    }

    return 0;
}

int WAV_ReadHeader(int fd, WAVContainer_t *container)
{
    assert((fd >=0) && container);

    if (read(fd, &container->header, sizeof(container->header)) != sizeof(container->header) ||
        read(fd, &container->format, sizeof(container->format)) != sizeof(container->format) ||
        read(fd, &container->chunk, sizeof(container->chunk)) != sizeof(container->chunk)) {

        fprintf(stderr, "Error WAV_ReadHeader/n");
        return -1;
    }

    if (WAV_P_CheckValid(container) < 0)
        return -1;

#ifdef WAV_PRINT_MSG
    WAV_P_PrintHeader(container);
#endif

    return 0;
}

int WAV_WriteHeader(int fd, WAVContainer_t *container)
{
    assert((fd >=0) && container);

    if (WAV_P_CheckValid(container) < 0)
        return -1;

    if (write(fd, &container->header, sizeof(container->header)) != sizeof(container->header) ||
        write(fd, &container->format, sizeof(container->format)) != sizeof(container->format) ||
        write(fd, &container->chunk, sizeof(container->chunk)) != sizeof(container->chunk)) {

        fprintf(stderr, "Error WAV_WriteHeader/n");
        return -1;
    }

#ifdef WAV_PRINT_MSG
    WAV_P_PrintHeader(container);
#endif

    return 0;
}

//File   : sndwav_common.h
//Author : Loon <[email protected]>

#ifndef __SNDWAV_COMMON_H
#define __SNDWAV_COMMON_H

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include "wav_parser.h"

typedef long long off64_t;

typedef struct SNDPCMContainer {
    snd_pcm_t *handle;
    snd_output_t *log;
    snd_pcm_uframes_t chunk_size;
    snd_pcm_uframes_t buffer_size;
    snd_pcm_format_t format;
    uint16_t channels;
    size_t chunk_bytes;
    size_t bits_per_sample;
    size_t bits_per_frame;

    uint8_t *data_buf;
} SNDPCMContainer_t;

ssize_t SNDWAV_ReadPcm(SNDPCMContainer_t *sndpcm, size_t rcount);

ssize_t SNDWAV_WritePcm(SNDPCMContainer_t *sndpcm, size_t wcount);

int SNDWAV_SetParams(SNDPCMContainer_t *sndpcm, WAVContainer_t *wav);
#endif /* #ifndef __SNDWAV_COMMON_H */

//File   : sndwav_common.c
//Author : Loon <[email protected]>

#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <alsa/asoundlib.h>

#include "sndwav_common.h"

int SNDWAV_P_GetFormat(WAVContainer_t *wav, snd_pcm_format_t *snd_format)
{
    if (LE_SHORT(wav->format.format) != WAV_FMT_PCM)
        return -1;

    switch (LE_SHORT(wav->format.sample_length)) {
    case 16:
        *snd_format = SND_PCM_FORMAT_S16_LE;
        break;
    case 8:
        *snd_format = SND_PCM_FORMAT_U8;
        break;
    default:
        *snd_format = SND_PCM_FORMAT_UNKNOWN;
        break;
    }

    return 0;
}

ssize_t SNDWAV_ReadPcm(SNDPCMContainer_t *sndpcm, size_t rcount)
{
    ssize_t r;
    size_t result = 0;
    size_t count = rcount;
    uint8_t *data = sndpcm->data_buf;

    if (count != sndpcm->chunk_size) {
        count = sndpcm->chunk_size;
    }

    while (count > 0) {
        r = snd_pcm_readi(sndpcm->handle, data, count);

        if (r == -EAGAIN || (r >= 0 && (size_t)r < count)) {
            snd_pcm_wait(sndpcm->handle, 1000);
        } else if (r == -EPIPE) {
            snd_pcm_prepare(sndpcm->handle);
            fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>/n");
        } else if (r == -ESTRPIPE) {
            fprintf(stderr, "<<<<<<<<<<<<<<< Need suspend >>>>>>>>>>>>>>>/n");
        } else if (r < 0) {
            fprintf(stderr, "Error snd_pcm_writei: [%s]", snd_strerror(r));
            exit(-1);
        }

        if (r > 0) {
            result += r;
            count -= r;
            data += r * sndpcm->bits_per_frame / 8;
        }
    }
    return rcount;
}

ssize_t SNDWAV_WritePcm(SNDPCMContainer_t *sndpcm, size_t wcount)
{
    ssize_t r;
    ssize_t result = 0;
    uint8_t *data = sndpcm->data_buf;

    if (wcount < sndpcm->chunk_size) {
        snd_pcm_format_set_silence(sndpcm->format,
            data + wcount * sndpcm->bits_per_frame / 8,
            (sndpcm->chunk_size - wcount) * sndpcm->channels);
        wcount = sndpcm->chunk_size;
    }
    while (wcount > 0) {
        r = snd_pcm_writei(sndpcm->handle, data, wcount);
        if (r == -EAGAIN || (r >= 0 && (size_t)r < wcount)) {
            snd_pcm_wait(sndpcm->handle, 1000);
        } else if (r == -EPIPE) {
            snd_pcm_prepare(sndpcm->handle);
            fprintf(stderr, "<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>/n");
        } else if (r == -ESTRPIPE) {
            fprintf(stderr, "<<<<<<<<<<<<<<< Need suspend >>>>>>>>>>>>>>>/n");
        } else if (r < 0) {
            fprintf(stderr, "Error snd_pcm_writei: [%s]", snd_strerror(r));
            exit(-1);
        }
        if (r > 0) {
            result += r;
            wcount -= r;
            data += r * sndpcm->bits_per_frame / 8;
        }
    }
    return result;
}

int SNDWAV_SetParams(SNDPCMContainer_t *sndpcm, WAVContainer_t *wav)
{
    snd_pcm_hw_params_t *hwparams;
    snd_pcm_format_t format;
    uint32_t exact_rate;
    uint32_t buffer_time, period_time;

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

    /* Init hwparams with full configuration space */
    if (snd_pcm_hw_params_any(sndpcm->handle, hwparams) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_any/n");
        goto ERR_SET_PARAMS;
    }

    if (snd_pcm_hw_params_set_access(sndpcm->handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_access/n");
        goto ERR_SET_PARAMS;
    }

    /* Set sample format */
    if (SNDWAV_P_GetFormat(wav, &format) < 0) {
        fprintf(stderr, "Error get_snd_pcm_format/n");
        goto ERR_SET_PARAMS;
    }
    if (snd_pcm_hw_params_set_format(sndpcm->handle, hwparams, format) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_format/n");
        goto ERR_SET_PARAMS;
    }
    sndpcm->format = format;

    /* Set number of channels */
    if (snd_pcm_hw_params_set_channels(sndpcm->handle, hwparams, LE_SHORT(wav->format.channels)) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_channels/n");
        goto ERR_SET_PARAMS;
    }
    sndpcm->channels = LE_SHORT(wav->format.channels);

    /* Set sample rate. If the exact rate is not supported */
    /* by the hardware, use nearest possible rate.         */
    exact_rate = LE_INT(wav->format.sample_rate);
    if (snd_pcm_hw_params_set_rate_near(sndpcm->handle, hwparams, &exact_rate, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_rate_near/n");
        goto ERR_SET_PARAMS;
    }
    if (LE_INT(wav->format.sample_rate) != exact_rate) {
        fprintf(stderr, "The rate %d Hz is not supported by your hardware./n ==> Using %d Hz instead./n",
            LE_INT(wav->format.sample_rate), exact_rate);
    }

    if (snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_get_buffer_time_max/n");
        goto ERR_SET_PARAMS;
    }
    if (buffer_time > 500000) buffer_time = 500000;
    period_time = buffer_time / 4;

    if (snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, hwparams, &buffer_time, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_buffer_time_near/n");
        goto ERR_SET_PARAMS;
    }

    if (snd_pcm_hw_params_set_period_time_near(sndpcm->handle, hwparams, &period_time, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params_set_period_time_near/n");
        goto ERR_SET_PARAMS;
    }

    /* Set hw params */
    if (snd_pcm_hw_params(sndpcm->handle, hwparams) < 0) {
        fprintf(stderr, "Error snd_pcm_hw_params(handle, params)/n");
        goto ERR_SET_PARAMS;
    }

    snd_pcm_hw_params_get_period_size(hwparams, &sndpcm->chunk_size, 0);
    snd_pcm_hw_params_get_buffer_size(hwparams, &sndpcm->buffer_size);
    if (sndpcm->chunk_size == sndpcm->buffer_size) {
        fprintf(stderr, ("Can't use period equal to buffer size (%lu == %lu)/n"), sndpcm->chunk_size, sndpcm->buffer_size);
        goto ERR_SET_PARAMS;
    }

    sndpcm->bits_per_sample = snd_pcm_format_physical_width(format);
    sndpcm->bits_per_frame = sndpcm->bits_per_sample * LE_SHORT(wav->format.channels);

    sndpcm->chunk_bytes = sndpcm->chunk_size * sndpcm->bits_per_frame / 8;

    /* Allocate audio data buffer */
    sndpcm->data_buf = (uint8_t *)malloc(sndpcm->chunk_bytes);
    if (!sndpcm->data_buf) {
        fprintf(stderr, "Error malloc: [data_buf]/n");
        goto ERR_SET_PARAMS;
    }

    return 0;

ERR_SET_PARAMS:
    return -1;
}

//File   : lplay.c
//Author : Loon <[email protected]>

#include <stdio.h>
#include <malloc.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <getopt.h>
#include <fcntl.h>
#include <ctype.h>
#include <errno.h>
#include <limits.h>
#include <time.h>
#include <locale.h>
#include <sys/unistd.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <alsa/asoundlib.h>
#include <assert.h>
#include "wav_parser.h"
#include "sndwav_common.h"

ssize_t SNDWAV_P_SaveRead(int fd, void *buf, size_t count)
{
    ssize_t result = 0, res;

    while (count > 0) {
        if ((res = read(fd, buf, count)) == 0)
            break;
        if (res < 0)
            return result > 0 ? result : res;
        count -= res;
        result += res;
        buf = (char *)buf + res;
    }
    return result;
}

void SNDWAV_Play(SNDPCMContainer_t *sndpcm, WAVContainer_t *wav, int fd)
{
    int load, ret;
    off64_t written = 0;
    off64_t c;
    off64_t count = LE_INT(wav->chunk.length);

    load = 0;
    while (written < count) {
        /* Must read [chunk_bytes] bytes data enough. */
        do {
            c = count - written;
            if (c > sndpcm->chunk_bytes)
                c = sndpcm->chunk_bytes;
            c -= load;

            if (c == 0)
                break;
            ret = SNDWAV_P_SaveRead(fd, sndpcm->data_buf + load, c);
            if (ret < 0) {
                fprintf(stderr, "Error safe_read/n");
                exit(-1);
            }
            if (ret == 0)
                break;
            load += ret;
        } while ((size_t)load < sndpcm->chunk_bytes);

        /* Transfer to size frame */
        load = load * 8 / sndpcm->bits_per_frame;
        ret = SNDWAV_WritePcm(sndpcm, load);
        if (ret != load)
            break;

        ret = ret * sndpcm->bits_per_frame / 8;
        written += ret;
        load = 0;
    }
}

int main(int argc, char *argv[])
{
    char *filename;
    char *devicename = "default";
    int fd;
    WAVContainer_t wav;
    SNDPCMContainer_t playback;

    if (argc != 2) {
        fprintf(stderr, "Usage: ./lplay <FILENAME>/n");
        return -1;
    }

    memset(&playback, 0x0, sizeof(playback));

    filename = argv[1];
    fd = open(filename, O_RDONLY);
    if (fd < 0) {
        fprintf(stderr, "Error open [%s]/n", filename);
        return -1;
    }

    if (WAV_ReadHeader(fd, &wav) < 0) {
        fprintf(stderr, "Error WAV_Parse [%s]/n", filename);
        goto Err;
    }

    if (snd_output_stdio_attach(&playback.log, stderr, 0) < 0) {
        fprintf(stderr, "Error snd_output_stdio_attach/n");
        goto Err;
    }

    if (snd_pcm_open(&playback.handle, devicename, SND_PCM_STREAM_PLAYBACK, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_open [ %s]/n", devicename);
        goto Err;
    }

    if (SNDWAV_SetParams(&playback, &wav) < 0) {
        fprintf(stderr, "Error set_snd_pcm_params/n");
        goto Err;
    }
    snd_pcm_dump(playback.handle, playback.log);

    SNDWAV_Play(&playback, &wav, fd);

    snd_pcm_drain(playback.handle);

    close(fd);
    free(playback.data_buf);
    snd_output_close(playback.log);
    snd_pcm_close(playback.handle);
    return 0;

Err:
    close(fd);
    if (playback.data_buf) free(playback.data_buf);
    if (playback.log) snd_output_close(playback.log);
    if (playback.handle) snd_pcm_close(playback.handle);
    return -1;
}

//File   : lrecord.c
//Author : Loon <[email protected]>

#include <stdio.h>
#include <malloc.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <getopt.h>
#include <fcntl.h>
#include <ctype.h>
#include <errno.h>
#include <limits.h>
#include <time.h>
#include <locale.h>
#include <sys/unistd.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <alsa/asoundlib.h>
#include <assert.h>
#include "wav_parser.h"
#include "sndwav_common.h"

#define DEFAULT_CHANNELS         (2)
#define DEFAULT_SAMPLE_RATE      (8000)
#define DEFAULT_SAMPLE_LENGTH    (16)
#define DEFAULT_DURATION_TIME    (10)

int SNDWAV_PrepareWAVParams(WAVContainer_t *wav)
{
    assert(wav);

    uint16_t channels = DEFAULT_CHANNELS;
    uint16_t sample_rate = DEFAULT_SAMPLE_RATE;
    uint16_t sample_length = DEFAULT_SAMPLE_LENGTH;
    uint32_t duration_time = DEFAULT_DURATION_TIME;

    /* Const */
    wav->header.magic = WAV_RIFF;
    wav->header.type = WAV_WAVE;
    wav->format.magic = WAV_FMT;
    wav->format.fmt_size = LE_INT(16);
    wav->format.format = LE_SHORT(WAV_FMT_PCM);
    wav->chunk.type = WAV_DATA;

    /* User definition */
    wav->format.channels = LE_SHORT(channels);
    wav->format.sample_rate = LE_INT(sample_rate);
    wav->format.sample_length = LE_SHORT(sample_length);

    /* See format of wav file */
    wav->format.blocks_align = LE_SHORT(channels * sample_length / 8);
    wav->format.bytes_p_second = LE_INT((uint16_t)(wav->format.blocks_align) * sample_rate);

    wav->chunk.length = LE_INT(duration_time * (uint32_t)(wav->format.bytes_p_second));
    wav->header.length = LE_INT((uint32_t)(wav->chunk.length) +/
        sizeof(wav->chunk) + sizeof(wav->format) + sizeof(wav->header) - 8);

    return 0;
}

void SNDWAV_Record(SNDPCMContainer_t *sndpcm, WAVContainer_t *wav, int fd)
{
    off64_t rest;
    size_t c, frame_size;

    if (WAV_WriteHeader(fd, wav) < 0) {
        exit(-1);
    }

    rest = wav->chunk.length;
    while (rest > 0) {
        c = (rest <= (off64_t)sndpcm->chunk_bytes) ? (size_t)rest : sndpcm->chunk_bytes;
        frame_size = c * 8 / sndpcm->bits_per_frame;
        if (SNDWAV_ReadPcm(sndpcm, frame_size) != frame_size)
            break;

        if (write(fd, sndpcm->data_buf, c) != c) {
            fprintf(stderr, "Error SNDWAV_Record[write]/n");
            exit(-1);
        }

        rest -= c;
    }
}

int main(int argc, char *argv[])
{
    char *filename;
    char *devicename = "default";
    int fd;
    WAVContainer_t wav;
    SNDPCMContainer_t record;

    if (argc != 2) {
        fprintf(stderr, "Usage: ./lrecord <FILENAME>/n");
        return -1;
    }

    memset(&record, 0x0, sizeof(record));

    filename = argv[1];
    remove(filename);
    if ((fd = open(filename, O_WRONLY | O_CREAT, 0644)) == -1) {
        fprintf(stderr, "Error open: [%s]/n", filename);
        return -1;
    }

    if (snd_output_stdio_attach(&record.log, stderr, 0) < 0) {
        fprintf(stderr, "Error snd_output_stdio_attach/n");
        goto Err;
    }

    if (snd_pcm_open(&record.handle, devicename, SND_PCM_STREAM_CAPTURE, 0) < 0) {
        fprintf(stderr, "Error snd_pcm_open [ %s]/n", devicename);
        goto Err;
    }

    if (SNDWAV_PrepareWAVParams(&wav) < 0) {
        fprintf(stderr, "Error SNDWAV_PrepareWAVParams/n");
        goto Err;
    }

    if (SNDWAV_SetParams(&record, &wav) < 0) {
        fprintf(stderr, "Error set_snd_pcm_params/n");
        goto Err;
    }
    snd_pcm_dump(record.handle, record.log);

    SNDWAV_Record(&record, &wav, fd);

    snd_pcm_drain(record.handle);

    close(fd);
    free(record.data_buf);
    snd_output_close(record.log);
    snd_pcm_close(record.handle);
    return 0;

Err:
    close(fd);
    remove(filename);
    if (record.data_buf) free(record.data_buf);
    if (record.log) snd_output_close(record.log);
    if (record.handle) snd_pcm_close(record.handle);
    return -1;
}

分享到：

下一篇：应用时间片实现多任务的单片机代码

查看评论

4楼 spiderget 2012-02-01 11:13发表 [回复]

您好，我把这些文件都保存下来了，然后用gcc编译 lplay.c 不成功，不知道是什么……在所在的文件夹下用这个语句
$ gcc -o lplay lplay.c -lasound
然后报错
/tmp/ccJTSnTU.o: In function `SNDWAV_Play':
lplay.c:(.text+0x19b): undefined reference to `SNDWAV_WritePcm'
/tmp/ccJTSnTU.o: In function `main':
lplay.c:(.text+0x2b9): undefined reference to `WAV_ReadHeader'
lplay.c:(.text+0x390): undefined reference to `SNDWAV_SetParams'
collect2: ld 返回 1

Re: sepnic 2012-02-01 11:39发表 [回复]

回复spiderget：因为你只是编译了lplay.c啊，没看到有好几个源文件吗？
gcc -o lplay lplay.c wav_parser.c sndwav_common.c -lasound
看看。
Linux下的编程，makefile编写都得要学习一下，如果被这些最基本的问题拌住，得不偿失。

Re: spiderget 2012-02-01 13:22发表 [回复]

回复sepnic：真是感谢啦！嗯，我会去学这些基本的东西的，不然又问这种小白问题……
谢谢及时回复！

3楼 liuchao35758600 2011-10-10 22:07发表 [回复]

这代码，有点长了。

Re: sepnic 2011-10-11 09:20发表 [回复]

回复liuchao35758600：应该算简单的吧，aplay的源码更复杂，当然它支持的功能也更多。我这个仅仅只是为了播放wav文件的，当时为了alsa而练手的东东。