mpv/libao2/ao_pcm.c

194 lines
4.3 KiB
C

#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),
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 */
fwrite(&wavhdr,sizeof(wavhdr),1,fp);
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);
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;
}