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mpv/libao2/ao_pcm.c
ulion 1ed6e96cfb Fix all current known multi-channel wrong order problems by adding
common functions for channel reordering.
This fixes these modules by adding channel reordering code for 5.0/5.1 audio:
ao: pcm
ad: dmo, faad, ffmpeg(ac3, dca, libfaad, liba52), pcm
ae: faac, lavc(ac3, libfaac), pcm


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@25343 b3059339-0415-0410-9bf9-f77b7e298cf2
2007-12-10 16:53:30 +00:00

233 lines
5.3 KiB
C

#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 "libaf/reorder_ch.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;
case AF_FORMAT_AC3:
bits=16;
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
if (ao_data.channels == 6 || ao_data.channels == 5) {
int frame_size = le2me_16(wavhdr.bits) / 8;
len -= len % (frame_size * ao_data.channels);
reorder_channel_nch(data, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
ao_data.channels,
len / frame_size, frame_size);
}
//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;
}