mirror of https://github.com/mpv-player/mpv
272 lines
7.3 KiB
C
272 lines
7.3 KiB
C
/*
|
|
* PCM audio output driver
|
|
*
|
|
* This file is part of MPlayer.
|
|
*
|
|
* MPlayer is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License as published by
|
|
* the Free Software Foundation; either version 2 of the License, or
|
|
* (at your option) any later version.
|
|
*
|
|
* MPlayer is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License along
|
|
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
|
|
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
|
|
*/
|
|
|
|
#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"
|
|
|
|
#ifdef __MINGW32__
|
|
// for GetFileType to detect pipes
|
|
#include <windows.h>
|
|
#endif
|
|
|
|
static const 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
|
|
#define WAV_ID_FLOAT_PCM 0x0003
|
|
|
|
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 uint64_t data_length;
|
|
|
|
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;
|
|
const 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");
|
|
}
|
|
|
|
if (ao_pcm_waveheader)
|
|
{
|
|
// WAV files must have one of the following formats
|
|
|
|
switch(format){
|
|
case AF_FORMAT_U8:
|
|
case AF_FORMAT_S16_LE:
|
|
case AF_FORMAT_S24_LE:
|
|
case AF_FORMAT_S32_LE:
|
|
case AF_FORMAT_FLOAT_LE:
|
|
case AF_FORMAT_AC3_BE:
|
|
case AF_FORMAT_AC3_LE:
|
|
break;
|
|
default:
|
|
format = AF_FORMAT_S16_LE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
bits = af_fmt2bits(format);
|
|
|
|
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(format == AF_FORMAT_FLOAT_LE ? WAV_ID_FLOAT_PCM : 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);
|
|
}
|
|
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){ /* Rewrite wave header */
|
|
int broken_seek = 0;
|
|
#ifdef __MINGW32__
|
|
// Windows, in its usual idiocy "emulates" seeks on pipes so it always looks
|
|
// like they work. So we have to detect them brute-force.
|
|
broken_seek = GetFileType((HANDLE)_get_osfhandle(_fileno(fp))) != FILE_TYPE_DISK;
|
|
#endif
|
|
if (broken_seek || fseek(fp, 0, SEEK_SET) != 0)
|
|
mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
|
|
else if (data_length > 0x7ffff000)
|
|
mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
|
|
else {
|
|
wavhdr.file_length = data_length + sizeof(wavhdr) - 8;
|
|
wavhdr.file_length = le2me_32(wavhdr.file_length);
|
|
wavhdr.data_length = le2me_32(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
|
|
//#if HAVE_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 == 5 || ao_data.channels == 6 || ao_data.channels == 8) {
|
|
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)
|
|
data_length += len;
|
|
|
|
return len;
|
|
}
|
|
|
|
// return: delay in seconds between first and last sample in buffer
|
|
static float get_delay(void){
|
|
|
|
return 0.0;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|