view libao2/ao_pcm.c @ 9660:c2d23e02522b

improvements to detc filter: -> use of 8x8 blocks rather than 16x16 to better localize the search for interlacing. this helps detect interlacing in very small motions, e.g. mouths in anime. -> removed some redundant conditions in the logic -> looser condition for detecting lacing and more forgiving of slight mismatches between fields from the two telecine frames to make up for quantization noise in low quality encodes. this code is still mostly experimental but probably better than the old version, so maybe it should be backported to 0.90...?

#include "config.h"

#include <stdio.h>
#include <stdlib.h>
#include <string.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),
        /* same conventions than in sox/wav.c/wavwritehdr() */
	0, //le2me_32(0x7ffff024),
	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),
	0, //le2me_32(0x7ffff000)
};

static FILE *fp = NULL;

// to set/get/query special features/parameters
static int control(int cmd,void *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 = strdup(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 */

	bits=8;
	switch(format){
	case AFMT_S8:
	    format=AFMT_U8;
	case AFMT_U8:
	    break;
	default:
	    format=AFMT_S16_LE;
	    bits=16;
	    break;
	}

	ao_data.outburst = 65536;
	ao_data.buffersize= 2*65536;
	ao_data.channels=channels;
	ao_data.samplerate=rate;
	ao_data.format=format;
	ao_data.bps=channels*rate*(bits/8);

	wavhdr.channels = le2me_16(ao_data.channels);
	wavhdr.sample_rate = le2me_32(ao_data.samplerate);
	wavhdr.bytes_per_second = le2me_32(ao_data.bps);
	wavhdr.bits = le2me_16(bits);
	
	wavhdr.data_length=le2me_32(0x7ffff000);
	wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;

	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 -vc null -vo null\n"
	       "PCM: Info: to write WAVE files use -waveheader (default); "
	       "for RAW PCM -nowaveheader.\n");

	fp = fopen(ao_outputfilename, "wb");
	if(fp) {
		if(ao_pcm_waveheader){ /* Reserve space for wave header */
			fwrite(&wavhdr,sizeof(wavhdr),1,fp);
			wavhdr.file_length=wavhdr.data_length=0;
		}
		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 && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */
		wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
		wavhdr.file_length = le2me_32(wavhdr.file_length);
		wavhdr.data_length = le2me_32(wavhdr.data_length);
		fwrite(&wavhdr,sizeof(wavhdr),1,fp);
	}
	fclose(fp);
}

// 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){

// let libaf to do the conversion...
#if 0
//#ifdef WORDS_BIGENDIAN
	if (ao_data.format == AFMT_S16_LE) {
	  unsigned short *buffer = (unsigned short *) data;
	  register int i;
	  for(i = 0; i < len/2; ++i) {
	    buffer[i] = le2me_16(buffer[i]);
	  }
	}
#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;
}