view libao2/ao_pcm.c @ 18971:ec2f6323fda3

Change SRC_PATH for ffmpeg back to '..' to avoid hardcoding current directory at configure time. This should work again now that libpostproc is no longer under libavcodec and all Makefiles included from ffmpeg are at the same directory level. The hardcoded paths caused breakage if the build directory was moved or copied after configure and prevented ccache from sharing compilation results between directories (different absolute include paths count as different compiler options).

#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;
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 = {
	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},
	  {"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.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(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;
}