view libao2/ao_pcm.c @ 23510:a6c619ee9d30

Teletext support for tv:// (v4l and v4l2 only) modified patch from Otvos Attila oattila at chello dot hu Module uses zvbi library for all low-level VBI operations (like I/O with vbi device, converting vbi pages into usefull vbi_page stuctures, rendering them into RGB32 images). All teletext related stuff (except properties, slave commands and rendering osd in text mode or RGB32 rendered teletext pages in spu mode) is implemented in tvi_vbi.c New properties: teletext_page - switching between pages teletext_mode - switch between on/off/opaque/transparent modes teletext_format - (currently read-only) allows to get format info (black/white,gray,text) teletext_half_page - trivial zooming (displaying top/bottom half of teletext page) New slave commands: teletext_add_dec - user interface for jumping to any page by editing page number interactively teletext_go_link - goes though links, specified on current page

#include "config.h"

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

#include "libavutil/common.h"
#include "mpbswap.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;
static int fast = 0;

#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;

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},
	  {"fast",       OPT_ARG_BOOL, &fast, NULL},
	  {NULL}
	};
	// set defaults
	ao_pcm_waveheader = 1;

	if (subopt_parse(ao_subdevice, subopts) != 0) {
	  return 0;
	}
	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 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.riff = le2me_32(WAV_ID_RIFF);
	wavhdr.wave = le2me_32(WAV_ID_WAVE);
	wavhdr.fmt = le2me_32(WAV_ID_FMT);
	wavhdr.fmt_length = le2me_32(16);
	wavhdr.fmt_tag = le2me_16(WAV_ID_PCM);
	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 = le2me_32(WAV_ID_DATA);
	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(void){

}

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

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

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

    if(vo_pts)
      return ao_data.pts < vo_pts + fast * 30000 ? 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(void){

    return 0.0;
}