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;
}