view libao2/ao_pcm.c @ 16946:47c5e9846cd3

ultra simple&slow pp filter, yes yet another spp like filter :) this one does actually compress&decompress the video at various shifts with lavc while the other spp filters are doing optimized intra only filtering limitations: mpeg4 is hardcoded, all options too, pretty trivial to change though, even filtering with non dct codecs like snow could be tried ... the qscale/qp is only taken fron the first MB of each image and then used for the whole image (would needs some small changes to lavc to let the user set the qscales for the mbs themselfs but iam to lazy ...) this needs ALOT of cpu time and memory especially at uspp=8 ...

#include "config.h"

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

#include "bswap.h"
#include "subopt-helper.h"
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "help_mp.h"


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

LIBAO_EXTERN(pcm)

extern int vo_pts;

static char *ao_outputfilename = NULL;
static 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
{
	uint32_t riff;
	uint32_t file_length;
	uint32_t wave;
	uint32_t fmt;
	uint32_t fmt_length;
	uint16_t fmt_tag;
	uint16_t channels;
	uint32_t sample_rate;
	uint32_t bytes_per_second;
	uint16_t block_align;
	uint16_t bits;
	uint32_t data;
	uint32_t 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;
	opt_t subopts[] = {
	  {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL},
	  {"file",       OPT_ARG_MSTRZ, &ao_outputfilename, NULL},
	  {NULL}
	};
	// set defaults
	ao_pcm_waveheader = 1;
	ao_outputfilename =
	      strdup((ao_pcm_waveheader)?"audiodump.wav":"audiodump.pcm");
	if (subopt_parse(ao_subdevice, subopts) != 0) {
	  return 0;
	}

	/* 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 AF_FORMAT_S8:
	    format=AF_FORMAT_U8;
	case AF_FORMAT_U8:
	    break;
	default:
	    format=AF_FORMAT_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.block_align = le2me_16(ao_data.channels * (bits / 8));
	
	wavhdr.data_length=le2me_32(0x7ffff000);
	wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;

	mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, 
	       (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, 
	       (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
	mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);

	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;
	}
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, 
               ao_outputfilename);
	return 0;
}

// close audio device
static void uninit(int immed){
	
	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);
	if (ao_outputfilename)
	  free(ao_outputfilename);
	ao_outputfilename = NULL;
}

// 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;
}