mpv/libao2/ao_pcm.c

#include <stdio.h>
#include <stdlib.h>

#include "config.h"
#include "bswap.h"
#include "afmt.h"
#include "audio_out.h"
#include "audio_out_internal.h"

static ao_info_t info = 
{
	"RAW PCM/WAVE file writer audio output",
	"pcm",
	"Atmosfear",
	""
};

LIBAO_EXTERN(pcm)

extern int vo_pts;

char *ao_outputfilename = NULL;
int ao_pcm_waveheader = 1;

#define WAV_ID_RIFF 0x46464952 /* "RIFF" */
#define WAV_ID_WAVE 0x45564157 /* "WAVE" */
#define WAV_ID_FMT  0x20746d66 /* "fmt " */
#define WAV_ID_DATA 0x61746164 /* "data" */
#define WAV_ID_PCM  0x0001

struct WaveHeader
{
	unsigned long riff;
	unsigned long file_length;
	unsigned long wave;
	unsigned long fmt;
	unsigned long fmt_length;
	short fmt_tag;
	short channels;
	unsigned long sample_rate;
	unsigned long bytes_per_second;
	short block_align;
	short bits;
	unsigned long data;
	unsigned long data_length;
};

/* init with default values */
static struct WaveHeader wavhdr = {
	le2me_32(WAV_ID_RIFF),
	le2me_32(0x00000000),
	le2me_32(WAV_ID_WAVE),
	le2me_32(WAV_ID_FMT),
	le2me_32(16),
	le2me_16(WAV_ID_PCM),
	le2me_16(2),
	le2me_32(44100),
	le2me_32(192000),
	le2me_16(4),
	le2me_16(16),
	le2me_32(WAV_ID_DATA),
	le2me_32(0x00000000)
};

static FILE *fp = NULL;

// to set/get/query special features/parameters
static int control(int cmd,int arg){
    return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
	int bits;
	if(!ao_outputfilename) {
		ao_outputfilename = (char *) malloc(sizeof(char) * 14);
		strcpy(ao_outputfilename,
		       (ao_pcm_waveheader ? "audiodump.wav" : "audiodump.pcm"));
	}
	
	/* bits is only equal to format if (format == 8) or (format == 16);
	   this means that the following "if" is a kludge and should
	   really be a switch to be correct in all cases */
	if (format == AFMT_S16_BE) { bits = 16;	}
	else { bits = format; }

	wavhdr.channels = le2me_16(channels);
	wavhdr.sample_rate = le2me_32(rate);
	wavhdr.bytes_per_second = rate * (bits / 8) * channels;
	wavhdr.bytes_per_second = le2me_32(wavhdr.bytes_per_second);
	wavhdr.bits = le2me_16(bits);

	printf("PCM: File: %s (%s)\n"
	       "PCM: Samplerate: %iHz Channels: %s Format %s\n",
	       ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
	       (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
	printf("PCM: Info: fastest dumping is achieved with -vo null "
	       "-hardframedrop.\n"
	       "PCM: Info: to write WAVE files use -waveheader (default); "
	       "for RAW PCM -nowaveheader.\n");

	fp = fopen(ao_outputfilename, "wb");

	ao_data.outburst = 65536;


	if(fp) {
		if(ao_pcm_waveheader) /* Reserve space for wave header */
			fseek(fp, sizeof(wavhdr), SEEK_SET);
		return 1;
	}
	printf("PCM: Failed to open %s for writing!\n", ao_outputfilename);
	return 0;
}

// close audio device
static void uninit(){
	
	if(ao_pcm_waveheader){ /* Write wave header */
		wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
		wavhdr.file_length = le2me_32(wavhdr.file_length);
		fseek(fp, 0, SEEK_SET);
		fwrite(&wavhdr,sizeof(wavhdr),1,fp);
	}
	fclose(fp);
	free(ao_outputfilename);
}

// stop playing and empty buffers (for seeking/pause)
static void reset(){

}

// stop playing, keep buffers (for pause)
static void audio_pause()
{
    // for now, just call reset();
    reset();
}

// resume playing, after audio_pause()
static void audio_resume()
{
}

// return: how many bytes can be played without blocking
static int get_space(){

    if(vo_pts)
      return ao_data.pts < vo_pts ? ao_data.outburst : 0;
    return ao_data.outburst;
}

// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){

#ifdef WORDS_BIGENDIAN
	register int i;
	unsigned short *buffer = (unsigned short *) data;

	if (wavhdr.bits == le2me_16(16)) {
	  for(i = 0; i < len/2; ++i) {
	    buffer[i] = le2me_16(buffer[i]);
	  }
	}
	/* FIXME: take care of cases with more than 8 bits here? */
#endif 
	
	//printf("PCM: Writing chunk!\n");
	fwrite(data,len,1,fp);

	if(ao_pcm_waveheader)
		wavhdr.data_length += len;
	
	return len;
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(){

    return 0.0;
}