mpv/libao2/ao_pcm.c

291 lines
8.2 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"
#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
#define WAV_ID_FORMAT_EXTENSIBLE 0xfffe
/* init with default values */
static uint64_t data_length;
static FILE *fp = NULL;
static void fput16le(uint16_t val, FILE *fp) {
uint8_t bytes[2] = {val, val >> 8};
fwrite(bytes, 1, 2, fp);
}
static void fput32le(uint32_t val, FILE *fp) {
uint8_t bytes[4] = {val, val >> 8, val >> 16, val >> 24};
fwrite(bytes, 1, 4, fp);
}
static void write_wave_header(FILE *fp, uint64_t data_length) {
int use_waveex = (ao_data.channels >= 5 && ao_data.channels <= 8);
uint16_t fmt = (ao_data.format == AF_FORMAT_FLOAT_LE) ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
int bits = af_fmt2bits(ao_data.format);
// Master RIFF chunk
fput32le(WAV_ID_RIFF, fp);
// RIFF chunk size: 'WAVE' + 'fmt ' + 4 + fmt_chunk_size + data chunk hdr (8) + data length
fput32le(12 + fmt_chunk_size + 8 + data_length, fp);
fput32le(WAV_ID_WAVE, fp);
// Format chunk
fput32le(WAV_ID_FMT, fp);
fput32le(fmt_chunk_size, fp);
fput16le(use_waveex ? WAV_ID_FORMAT_EXTENSIBLE : fmt, fp);
fput16le(ao_data.channels, fp);
fput32le(ao_data.samplerate, fp);
fput32le(ao_data.bps, fp);
fput16le(ao_data.channels * (bits / 8), fp);
fput16le(bits, fp);
if (use_waveex) {
// Extension chunk
fput16le(22, fp);
fput16le(bits, fp);
switch (ao_data.channels) {
case 5:
fput32le(0x0607, fp); // L R C Lb Rb
break;
case 6:
fput32le(0x060f, fp); // L R C Lb Rb LFE
break;
case 7:
fput32le(0x0727, fp); // L R C Cb Ls Rs LFE
break;
case 8:
fput32le(0x063f, fp); // L R C Lb Rb Ls Rs LFE
break;
}
// 2 bytes format + 14 bytes guid
fput32le(fmt, fp);
fput32le(0x00100000, fp);
fput32le(0xAA000080, fp);
fput32le(0x719B3800, fp);
}
// Data chunk
fput32le(WAV_ID_DATA, fp);
fput32le(data_length, fp);
}
// 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){
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;
}
}
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*(af_fmt2bits(format)/8);
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
(ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
(channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] Info: Faster dumping is achieved with -vc null -vo null -ao pcm:fast\n[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
fp = fopen(ao_outputfilename, "wb");
if(fp) {
if(ao_pcm_waveheader){ /* Reserve space for wave header */
write_wave_header(fp, 0x7ffff000);
}
return 1;
}
mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
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 > 0xfffff000) {
mp_msg(MSGT_AO, MSGL_ERR, "File larger than allowed for WAV files, may play truncated!\n");
data_length = 0xfffff000;
}
write_wave_header(fp, data_length);
}
}
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 = af_fmt2bits(ao_data.format) / 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;
}